Chapter 1 Voip Configuration Commands 1.1 Voip

Transcript

Command Manual – Voice Comware V3 Table of Contents Table of Contents Chapter 1 VoIP Configuration Commands ................................................................................. 1-1 1.1 VoIP Configuration Commands ......................................................................................... 1-1 1.1.1 address.................................................................................................................... 1-1 1.1.2 area ......................................................................................................................... 1-2 1.1.3 area-id (voice entity view) ....................................................................................... 1-3 1.1.4 busytone-t-th ........................................................................................................... 1-3 1.1.5 caller-permit............................................................................................................. 1-4 1.1.6 cid display................................................................................................................ 1-6 1.1.7 cid enable ................................................................................................................ 1-7 1.1.8 cid send ................................................................................................................... 1-7 1.1.9 cid type .................................................................................................................... 1-8 1.1.10 cng-on ................................................................................................................... 1-9 1.1.11 compression ........................................................................................................ 1-10 1.1.12 cptone.................................................................................................................. 1-16 1.1.13 debugging voice cm ............................................................................................ 1-19 1.1.14 debugging voice data-flow .................................................................................. 1-20 1.1.15 debugging voice dpl ............................................................................................ 1-20 1.1.16 debugging voice h225 ......................................................................................... 1-21 1.1.17 debugging voice h245 ......................................................................................... 1-21 1.1.18 debugging voice ipp ............................................................................................ 1-22 1.1.19 debugging voice rcv ............................................................................................ 1-23 1.1.20 debugging voice vas ........................................................................................... 1-23 1.1.21 debugging voice vmib ......................................................................................... 1-24 1.1.22 debugging voice vpp ........................................................................................... 1-25 1.1.23 default entity compression .................................................................................. 1-26 1.1.24 default entity normal-connect slow-h245 ............................................................ 1-27 1.1.25 default entity payload-size................................................................................... 1-27 1.1.26 default entity service data enable........................................................................ 1-28 1.1.27 default entity service data payload-size .............................................................. 1-29 1.1.28 default entity vad-on............................................................................................ 1-30 1.1.29 delay.................................................................................................................... 1-30 1.1.30 delay-reversal...................................................................................................... 1-31 1.1.31 description (voice entity view) ............................................................................. 1-32 1.1.32 description (voice subscriber line view) .............................................................. 1-32 1.1.33 dial-prefix............................................................................................................. 1-33 1.1.34 dial-program ........................................................................................................ 1-35 1.1.35 display voice call-history-record .......................................................................... 1-35 1.1.36 display voice call-info .......................................................................................... 1-37 i Command Manual – Voice Comware V3 Table of Contents 1.1.37 display voice default............................................................................................ 1-38 1.1.38 display voice entity .............................................................................................. 1-40 1.1.39 display voice ipp .................................................................................................. 1-41 1.1.40 display voice number-substitute.......................................................................... 1-43 1.1.41 display voice rcv ccb ........................................................................................... 1-43 1.1.42 display voice rcv statistic..................................................................................... 1-46 1.1.43 display voice sip register-state ............................................................................ 1-49 1.1.44 display voice subscriber-line ............................................................................... 1-49 1.1.45 display voice voip data-statistic........................................................................... 1-52 1.1.46 display voice vpp ................................................................................................. 1-54 1.1.47 dot-match ............................................................................................................ 1-57 1.1.48 dscp media .......................................................................................................... 1-58 1.1.49 dtmf sensitivity-level ............................................................................................ 1-59 1.1.50 dtmf threshold...................................................................................................... 1-59 1.1.51 echo-canceller ..................................................................................................... 1-64 1.1.52 em-phy-parm ....................................................................................................... 1-65 1.1.53 em-signal............................................................................................................. 1-66 1.1.54 entity.................................................................................................................... 1-67 1.1.55 fast-connect......................................................................................................... 1-68 1.1.56 first-rule ............................................................................................................... 1-69 1.1.57 first-rule ............................................................................................................... 1-70 1.1.58 hookoff-time......................................................................................................... 1-70 1.1.59 impedance........................................................................................................... 1-71 1.1.60 line....................................................................................................................... 1-72 1.1.61 match-template.................................................................................................... 1-73 1.1.62 max-call (voice dial program view)...................................................................... 1-75 1.1.63 max-call (voice entity view) ................................................................................. 1-76 1.1.64 normal-connect slow-h245 .................................................................................. 1-77 1.1.65 number-match ..................................................................................................... 1-78 1.1.66 number-substitute ............................................................................................... 1-79 1.1.67 open-trunk ........................................................................................................... 1-79 1.1.68 outband ............................................................................................................... 1-80 1.1.69 overlap timer........................................................................................................ 1-82 1.1.70 overlap voip h323................................................................................................ 1-82 1.1.71 payload-size ........................................................................................................ 1-83 1.1.72 plc-mode.............................................................................................................. 1-85 1.1.73 priority.................................................................................................................. 1-85 1.1.74 private-line........................................................................................................... 1-86 1.1.75 progress-tone ...................................................................................................... 1-87 1.1.76 receive gain ......................................................................................................... 1-88 1.1.77 reset voice call-history-record line....................................................................... 1-89 1.1.78 reset voice ipp ..................................................................................................... 1-89 ii Command Manual – Voice Comware V3 Table of Contents 1.1.79 reset voice rcv ..................................................................................................... 1-90 1.1.80 reset voice vpp .................................................................................................... 1-90 1.1.81 reset voice voip data-statistic .............................................................................. 1-91 1.1.82 rule ...................................................................................................................... 1-91 1.1.83 select-rule rule-order ........................................................................................... 1-95 1.1.84 select-rule search-stop........................................................................................ 1-96 1.1.85 select-rule type-first............................................................................................. 1-97 1.1.86 select-stop ........................................................................................................... 1-98 1.1.87 send-busytone..................................................................................................... 1-99 1.1.88 send-number (voice entity view) ......................................................................... 1-99 1.1.89 send-number (voice subscriber-line view) ........................................................ 1-100 1.1.90 send-ring ........................................................................................................... 1-101 1.1.91 service data enable ........................................................................................... 1-101 1.1.92 shutdown (voice entity view) ............................................................................. 1-102 1.1.93 shutdown (voice subscriber-line view) .............................................................. 1-102 1.1.94 silence-th-span .................................................................................................. 1-103 1.1.95 special-service................................................................................................... 1-104 1.1.96 subscriber-line ................................................................................................... 1-105 1.1.97 substitute (voice subscriber line/entity voip/entity vofr view) ............................ 1-105 1.1.98 substitute (voice dial-program/pots entity view) ................................................ 1-106 1.1.99 terminator .......................................................................................................... 1-108 1.1.100 timer dial-interval............................................................................................. 1-108 1.1.101 timer first-dial................................................................................................... 1-109 1.1.102 timer ring-back................................................................................................. 1-110 1.1.103 timer wait-digit ................................................................................................. 1-110 1.1.104 trace interval.................................................................................................... 1-111 1.1.105 transmit gain.................................................................................................... 1-112 1.1.106 tunnel-on ......................................................................................................... 1-113 1.1.107 type.................................................................................................................. 1-113 1.1.108 vad-on ............................................................................................................. 1-114 1.1.109 vi-card busy-tone-detect.................................................................................. 1-115 1.1.110 vi-card cptone-custom..................................................................................... 1-116 1.1.111 vi-card reboot .................................................................................................. 1-117 1.1.112 voice-setup ...................................................................................................... 1-118 1.1.113 voip calledtunnel ............................................................................................. 1-119 1.1.114 voip call-start ................................................................................................... 1-119 1.1.115 voip h323-descriptor........................................................................................ 1-120 1.1.116 voip timer......................................................................................................... 1-121 1.1.117 vqa data-statistic ............................................................................................. 1-121 1.1.118 vqa dscp .......................................................................................................... 1-122 1.1.119 vqa dsp-monitor .............................................................................................. 1-124 1.1.120 vqa performance ............................................................................................. 1-125 iii Command Manual – Voice Comware V3 Table of Contents Chapter 2 BSV Configuration Commands .................................................................................. 2-1 2.1 BSV Configuration Commands.......................................................................................... 2-1 2.1.1 permanent-active..................................................................................................... 2-1 2.1.2 power-source........................................................................................................... 2-1 Chapter 3 VoFR Configuration Commands ................................................................................ 3-1 3.1 VoFR Configuration Commands........................................................................................ 3-1 3.1.1 address.................................................................................................................... 3-1 3.1.2 call-mode................................................................................................................. 3-2 3.1.3 cid select-mode ....................................................................................................... 3-3 3.1.4 debugging voice vofr ............................................................................................... 3-3 3.1.5 display fr vofr-info .................................................................................................... 3-4 3.1.6 display voice vofr call .............................................................................................. 3-5 3.1.7 display voice vofr statistic........................................................................................ 3-8 3.1.8 motorola base-svc................................................................................................... 3-9 3.1.9 motorola encapsulation ......................................................................................... 3-10 3.1.10 motorola max-voice............................................................................................. 3-11 3.1.11 motorola remote-id .............................................................................................. 3-11 3.1.12 outband vofr ........................................................................................................ 3-12 3.1.13 priority.................................................................................................................. 3-12 3.1.14 send-called number............................................................................................. 3-13 3.1.15 seq-number ......................................................................................................... 3-14 3.1.16 timestamp............................................................................................................ 3-15 3.1.17 trunk-id ................................................................................................................ 3-15 3.1.18 vad-on ................................................................................................................. 3-16 3.1.19 vofr ...................................................................................................................... 3-17 3.1.20 vofr frf11-timer ..................................................................................................... 3-18 3.1.21 vofr jitter-buffer .................................................................................................... 3-19 3.1.22 voice bandwidth .................................................................................................. 3-19 Chapter 4 E1/T1 Voice Configuration Commands ..................................................................... 4-1 4.1 E1/T1 Voice Configuration Commands ............................................................................. 4-1 4.1.1 ani............................................................................................................................ 4-1 4.1.2 ani-offset.................................................................................................................. 4-2 4.1.3 answer..................................................................................................................... 4-2 4.1.4 cas........................................................................................................................... 4-3 4.1.5 clear-forward-ack..................................................................................................... 4-4 4.1.6 debugging voice r2.................................................................................................. 4-5 4.1.7 debugging voice rcv r2 ............................................................................................ 4-6 4.1.8 debugging voice vpp r2 ........................................................................................... 4-6 4.1.9 default...................................................................................................................... 4-7 4.1.10 delay...................................................................................................................... 4-9 4.1.11 display voice em call-statistic .............................................................................. 4-11 4.1.12 display voice em ccb ........................................................................................... 4-13 iv Command Manual – Voice Comware V3 Table of Contents 4.1.13 display voice r2 call-statistics .............................................................................. 4-14 4.1.14 display voice rcv statistic r2................................................................................. 4-16 4.1.15 display voice subscriber-line ............................................................................... 4-18 4.1.16 display voice voip ................................................................................................ 4-20 4.1.17 dl-bits................................................................................................................... 4-21 4.1.18 dtmf ..................................................................................................................... 4-22 4.1.19 effect-time............................................................................................................ 4-23 4.1.20 final-callednum .................................................................................................... 4-24 4.1.21 force-metering ..................................................................................................... 4-24 4.1.22 group-b ................................................................................................................ 4-25 4.1.23 line....................................................................................................................... 4-26 4.1.24 loopback .............................................................................................................. 4-27 4.1.25 mfc (R2 CAS) ...................................................................................................... 4-27 4.1.26 mode ................................................................................................................... 4-28 4.1.27 open-trunk ........................................................................................................... 4-30 4.1.28 pcm...................................................................................................................... 4-31 4.1.29 pri-set .................................................................................................................. 4-32 4.1.30 re-answer ............................................................................................................ 4-33 4.1.31 register-number ................................................................................................... 4-34 4.1.32 register-value....................................................................................................... 4-35 4.1.33 renew................................................................................................................... 4-38 4.1.34 reset voice em ..................................................................................................... 4-39 4.1.35 reset voice r2....................................................................................................... 4-40 4.1.36 reverse ................................................................................................................ 4-40 4.1.37 seizure-ack .......................................................................................................... 4-41 4.1.38 select-mode......................................................................................................... 4-42 4.1.39 send-dialtone....................................................................................................... 4-43 4.1.40 sendring............................................................................................................... 4-43 4.1.41 signal-value ......................................................................................................... 4-44 4.1.42 special-character ................................................................................................. 4-45 4.1.43 subscriber-line ..................................................................................................... 4-46 4.1.44 timer (digital E&M)............................................................................................... 4-47 4.1.45 timer dtmf (R2) .................................................................................................... 4-48 4.1.46 timer register-pulse (R2) ..................................................................................... 4-49 4.1.47 timer register-complete (R2) ............................................................................... 4-49 4.1.48 timer ring (R2) ..................................................................................................... 4-50 4.1.49 timer dl (R2)......................................................................................................... 4-51 4.1.50 timeslot-set .......................................................................................................... 4-52 4.1.51 trunk-direction...................................................................................................... 4-54 4.1.52 ts.......................................................................................................................... 4-55 Chapter 5 Fax Configuration Commands ................................................................................... 5-1 5.1 Fax Configuration Commands ........................................................................................... 5-1 v Command Manual – Voice Comware V3 Table of Contents 5.1.1 cngced-detection ..................................................................................................... 5-1 5.1.2 debugging voice fax ................................................................................................ 5-2 5.1.3 default entity fax ...................................................................................................... 5-3 5.1.4 default entity modem compatible-param................................................................. 5-5 5.1.5 default entity modem protocol ................................................................................. 5-6 5.1.6 display voice fax ...................................................................................................... 5-7 5.1.7 fax baudrate ............................................................................................................ 5-9 5.1.8 fax ecm.................................................................................................................. 5-10 5.1.9 fax level ................................................................................................................. 5-11 5.1.10 fax nsf-on............................................................................................................. 5-12 5.1.11 fax protocol.......................................................................................................... 5-13 5.1.12 fax support-mode ................................................................................................ 5-14 5.1.13 fax train-mode ..................................................................................................... 5-15 5.1.14 modem compatible-param .................................................................................. 5-16 5.1.15 modem protocol................................................................................................... 5-16 5.1.16 reset voice fax statistics ...................................................................................... 5-17 5.1.17 reset voice fax trans-statistics ............................................................................. 5-18 5.1.18 voip h323-conf tcs-t38......................................................................................... 5-18 Chapter 6 Voice RADIUS Configuration Commands ................................................................. 6-1 6.1 Voice RADIUS Configuration Commands ......................................................................... 6-1 6.1.1 aaa-client................................................................................................................. 6-1 6.1.2 authentication .......................................................................................................... 6-1 6.1.3 authentication-did .................................................................................................... 6-2 6.1.4 callednumber receive-method................................................................................. 6-3 6.1.5 card-digit.................................................................................................................. 6-4 6.1.6 cdr ........................................................................................................................... 6-4 6.1.7 delay receive-dial .................................................................................................... 6-6 6.1.8 debugging voice vcc................................................................................................ 6-6 6.1.9 display aaa unsent-h323-call-record ....................................................................... 6-8 6.1.10 display voice aaa-client configuration ................................................................... 6-9 6.1.11 display voice call-history-record .......................................................................... 6-10 6.1.12 display voice vcc ................................................................................................. 6-13 6.1.13 gw-access-number.............................................................................................. 6-16 6.1.14 password-digit ..................................................................................................... 6-17 6.1.15 process-config ..................................................................................................... 6-18 6.1.16 redialtimes ........................................................................................................... 6-20 6.1.17 reset voice vcc..................................................................................................... 6-21 6.1.18 selectlanguage .................................................................................................... 6-21 Chapter 7 GK Client Configuration Commands ......................................................................... 7-1 7.1 GK Client Configuration commands .................................................................................. 7-1 7.1.1 area-id (in Voice GK Client View)............................................................................ 7-1 7.1.2 debugging voice ras ................................................................................................ 7-2 vi Command Manual – Voice Comware V3 Table of Contents 7.1.3 display voice gateway ............................................................................................. 7-2 7.1.4 gk-client ................................................................................................................... 7-3 7.1.5 gk-2nd-id ................................................................................................................. 7-4 7.1.6 gk-id......................................................................................................................... 7-5 7.1.7 gk-security call......................................................................................................... 7-5 7.1.8 gk-security register-pwd .......................................................................................... 7-6 7.1.9 gw-address.............................................................................................................. 7-7 7.1.10 gw-id...................................................................................................................... 7-8 7.1.11 ras-on .................................................................................................................... 7-8 Chapter 8 SIP Client Commands ................................................................................................. 8-1 8.1 SIP Client Commands ....................................................................................................... 8-1 8.1.1 address sip .............................................................................................................. 8-1 8.1.2 debugging voice sip ................................................................................................ 8-2 8.1.3 display voice sip call-statistics................................................................................. 8-2 8.1.4 mode (SIP client)..................................................................................................... 8-6 8.1.5 register-enable ........................................................................................................ 8-7 8.1.6 reset voice sip ......................................................................................................... 8-8 8.1.7 sip............................................................................................................................ 8-8 8.1.8 sip-call forwarding ................................................................................................... 8-9 8.1.9 sip-comp.................................................................................................................. 8-9 8.1.10 sip-comp agent.................................................................................................... 8-10 8.1.11 sip-domain........................................................................................................... 8-11 8.1.12 sip-id.................................................................................................................... 8-11 8.1.13 sip-server............................................................................................................. 8-12 8.1.14 source-ip.............................................................................................................. 8-13 8.1.15 wildcard-register enable ...................................................................................... 8-14 vii Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Chapter 1 VoIP Configuration Commands 1.1 VoIP Configuration Commands 1.1.1 address Syntax address { ip ip-address | ras } undo address { ip | ras } View Voice entity view Parameter ip ip-address: Indicate a VoIP dial entity session destination, i.e. the called IP address. ras: Router uses RAS recommendation to interact information with GK Server to map the called phone number to the IP address of peer voice gateway. It is used only in the networking configuration that uses GK (gatekeeper) to provide voice IP services. Description Use the address command to configure the voice routing policy to the peer voice gateway. Use the undo address command to cancel the voice routing policy that has been configured. By default, no routing policy is configured. This command is used to configure the network address for the VolP voice entity. The system supports the following two VolP routing policies at present. z Static routing policy: Find the IP address of destination voice gateway in static mode according to address ip ip-address command. z Dynamic routing policy: The router and GK Server interact with RAS information after the address ras command is configured. GK will dynamically send back the peer voice gateway address that matches the called number to the router. Related command: address sip, match-template. Example # Configure the destination IP address corresponding to the called 12345 as 10.1.1.2. [H3C-voice-dial-entity1] match-template 12345 [H3C-voice-dial-entity1] address ip 10.1.1.2 1-1 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.2 area Syntax area { north-america | custom | europe } undo area View Voice subscriber-line view Parameter north-america: Busy tone type of the switch connected to this subscriber-line is of North America standard. custom: Busy tone type of the switch connected to this subscriber-line is defined by the users. europe: Busy tone type of the switch connected to this subscriber-line is of Europe standard. Description Use the area command to configure the type of busy tone detection for FXO voice subscriber-line. Use the undo area command to restore the default value. By default, europe busy tone type standard is set. This command is used only for 2-wire loop trunk subscriber-line FXO, and it can only perform configuration to the first voice subscriber-line on the voice card. If successful, the configuration will be effective for all the voice subscriber-lines of the voice card. When this subscriber-line is connected to a common user line of a program-controlled switch, if the user on the switch side hooks on first, only by detecting the busy tone can the router know the user on-hooking operation. Since different switches execute different prompt tone schemes, there exist different frequency spectrum characteristics. This command is used to set the frequency spectrum characteristic used by the router to detect the existence of the busy tone. Example # Use north-america standard to detect the existence of the busy tone on voice subscriber-line 0/0/1. [H3C-voice-line0/0/1] area north-america 1-2 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.3 area-id (voice entity view) Syntax area-id string undo area-id View Voice entity view Parameter string: Area ID, an integer in the range of 0 to 9. The “#” can be used. Description Use the area-id command to configure the area ID of voice GW. Use the undo area-id command to cancel the specified area ID. By default, no area ID of voice GW is configured. The voice area ID is set in VoIP voice entity view and will be automatically added to the beginning of called numbers when establishing calls. Related command: match-template, entity. Example # Configure the VoIP voice entity 101 with the area ID 6#. [H3C-voice-dial-entity101] area-id 6# 1.1.4 busytone-t-th Syntax busytone-t-th time-threshold undo busytone-t-th View FXO voice subscriber-line view Parameter time-threshold: Threshold of busy tone detection. It ranges from 2 to 12, with a bigger value meaning longer detection time. The threshold defaults to 2, that is, the device hangs up upon two contiguous detections of busy tone. Description Use the busytone-t-th command to configure the threshold of busy tone detection. 1-3 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Use the undo busytone-t-th command to restore the default. The actual busy tone data does not always match the configured parameter. If the difference is large, detection inaccuracy may occur, resulting in on-hook failure or improper on-hook. You can however, tune the threshold of busy tone detection to achieve detection accuracy. For example, you can eliminate improper on-hooks caused by busy tone data inaccuracy by increasing the time for busy tone detection. Note that before you configure a threshold of busy tone detection, you must test it fully making sure that on-hook operation can be done properly. Example # Set the threshold of busy tone detection to 3. [H3C-voice-line0] busytone-t-th 3 1.1.5 caller-permit Syntax caller-permit calling-string undo caller-permit { calling-string | all } View Voice entity view Parameter all: All callers. calling-string: Calling numbers that are permitted to call in, in the format of { [ + ] string [ $ ]| $ }. The largest length of the string is 31. The symbols are described in the following: +: Appears at the beginning of a calling number to indicate that the number is z E.164-compliant. $: As the last character to indicate the end of the number. That means the entire z calling number must match all the characters before “$” in the string. If there is only “$” in the string, the calling number can be empty. string: A string composed of any characters of “0123456789ABCD#*.!+%[]()-”. z The meanings of the characters are described in the following table: Table 1-1 Meanings of the characters in string Character Meaning 0-9 Numbers from 0 to 9. Each means a digit. ABCD Each character means a digit. # and * Each means a valid digit. 1-4 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Character Meaning . A wildcard. It can match any digit of a valid number. For example, 555. . . . matches any string that begins with 555 and with four additional characters. ! The character or characters right in front of it does not appear or appears once. For example, 56!1234 can match 51234 and 561234. + The character or characters right in front of it appears once or several times. But its appearance at the beginning of the whole number means the number is E.164-compliant. - Hyphen. It connects two values (the smaller one before it and the bigger one after it) to indicate a range. For example, “1-9” means numbers from 1 to 9 (inclusive). % The character or characters right in front of it does not appear, or appears once or several times. [] Selects one character from the group. For example, [1-36A] can match only one character among 1, 2, 3, 6, and A. () A group of characters. For example, (123A) means a string “123A”. 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 at least appear once. Note: z The character or characters in front of "!”, “%”, and “+” are not to be matched accurately. They are handled similar to the wildcard “.”. Moreover, these symbols cannot be used alone. There must be a valid digit or digits in front of them. z If you want to use “[ ]” and “( )” at the same time, you must use them in the format “( [ ] )”. Other formats, such as “[ [ ] ]” and “[ ( ) ]” are illegal. z “-“ can only be used inside “[ ]”, and it only connects the same type of characters, such as “0-9”. The formats like “0-A” are illegal. Description Use the caller-permit command to configure the calling numbers that are permitted to call in. Use the undo caller-permit command to delete the calling numbers that are permitted to call in. By default, no calling number is configured. That means there is no limitation on calling numbers. 1-5 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands You can configure 32 calling numbers for a voice entity at most. If you only use “$”, empty calling numbers are permitted to call in. Related command: match-template. Example # Configure voice entity 2 to permit 660268 or empty calling numbers to call in. [H3C-voice-dial-entity2] caller-permit 660268$ [H3C-voice-dial-entity2] caller-permit $ # Configure voice entity 2 to permit the calling numbers beginning with 20 to call in. [H3C-voice-dial-entity2] caller-permit 20 1.1.6 cid display Syntax cid display undo cid display View Voice subscriber-line view Parameter None Description Use the cid display command to enable caller identification display. Use the undo cid display command to disable caller identification display. By default, caller identification display is enabled. This command is applicable to FXS subscriber-lines. When functioning as the called, the FXS module can send caller identification information to its called phone between the first and second rings. When disabled to send caller identification information, the FXS interface sends the character “P” received from the IP side instead. Thus, the called phone is unable to display caller identification information. Example # Enable caller identification display on voice subscriber-line 1/0/0. [H3C-voice-line1/0/0] cid display 1-6 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.7 cid enable Syntax cid enable undo cid enable View Voice subscriber-line view Parameter None Description Use the cid enable command to enable CID on the FXO interface. Use the undo cid enable command to disable CID on the FXO interface. By default, CID is enabled on the FXO interface. This command applies to FXO voice subscriber-lines only. With CID enabled, the FXO interface can receive the modulated caller identification data from an analog line between the first ring and second rings and then send the data demodulated with FSK to the IP side. With CID disabled, the local FXO interface does the following when the calling party sends a calling number: z If a number is configured in the match template for the POTS entity associated with the local FXO interface, the interface substitutes this number for the calling number and sends it to the called side. z If wildcard dots (.) are used in the number configured in the match template for the POTS entity associated with the local FXO interface, the interface substitutes zeros for the calling number’s digits in the place of dots, for example, 1000 for 1… and then sends the substitution number to the called side. Example # Enable CID on FXO voice subscriber-line 1/0/0. [H3C-voice-line1/0/0] cid enable 1.1.8 cid send Syntax cid send undo cid send 1-7 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View Voice subscriber-line view Parameter None Description Use the cid send command to enable the FXO or FXS module to send calling numbers to the IP side. Use the undo cid send command to disable the FXO or FXS module to send calling numbers to the IP side. By default, calling numbers are sent to the IP side. This command applies to FXS and FXO subscriber-lines only. After you configure the undo cid send command, the FXO interface does not send any number to the called side, regardless of whether the calling party has sent a calling number and regardless of whether a number is configured in the match template for the voice entity associated with the FXO interface. Example # Enable the FXO voice subscriber-line 3/0/0 to send calling numbers to the IP side. [H3C-voice-line3/0/0] cid send # Disable the FXS voice subscriber-line 1/0/0 to send calling numbers to the IP side. [H3C-voice-line1/0/0] undo cid send 1.1.9 cid type Syntax cid type { complex | simple } View Voice subscriber-line view Parameter complex: Caller identification information is transmitted in multiple-data message format (MDMF). simple: Caller identification information is transmitted in single-data message format (SDMF). 1-8 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the cid type command to configure the format of transmitted information about the calling party. Two formats are available: multiple data message format (MDMF) and single data message format (SDMF). When the remote end supports one format only, you must use the same setting at the local end. This command applies to both FXO and FXS subscriber-lines. Example # Set the format of the transmitted caller identification information to SDMF on voice subscriber line 1/0/0. [H3C-voice-line1/0/0] cid type simple 1.1.10 cng-on Syntax cng-on undo cng-on View Voice subscriber-line view Parameter None Description Use the cng-on command to enable comfort noise function. Use the undo cng-on command to disable the comfort noise function,. By default, comfort noise setting is enabled. This command is applicable to FXO, FXS, E&M subscriber-lines and digital E1 voice subscriber-line. When the silence detecting function on a corresponding voice entity is enabled, some background noise can be generated by using the command to fill the toneless intervals during a conversation. If no comfort noise is generated, the toneless intervals during a conversation will cause the interlocutors uncomfortable. Related command: subscriber-line and vad-on. Example # Disable comfort noise function on subscriber line 1/0/0. [H3C-voice-line1/0/0] undo cng-on 1-9 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.11 compression Syntax compression { 1st-level | 2nd-level | 3rd-level | 4th-level } { g711alaw | g711ulaw | g723r53 | g723r63 | g729a | g729r8 | g729ab | g729br8 | g726r16 | g726r24 | g726r32 | g726r40 } undo compression { 1st-level | 2nd-level | 3rd-level | 4th-level } View Voice entity view Parameter 1st-level: Indicates the first selected voice compression method. 2nd-level: Indicates the second selected voice compression method. 3rd-level: Indicates the third selected voice compression method. 4th-level: Indicates the fourth selected voice compression method. g711alaw: Specifies G.711 A-law coding (defining the pulse code modulation technology), requiring the bandwidth of 64 kbps, usually adopted by Europe. g711ulaw: Specifies G.711μ-law coding, requiring the bandwidth of 64 kbps, usually adopted in the North America and Japan. g723r53: Specifies G.723.1 Annex A coding, requiring the bandwidth of 5.3 kbps. g723r63: Specifies G.723.1 Annex A coding, requiring the bandwidth of 6.3 kbps. g729a: Specifies G.729 Annex A coding (a simplified version of G.729 coding), requiring the bandwidth of 8 kbps. g729r8: Specifies G.729 (the voice coding technology using conjugate algebraic-code-excited linear-prediction) coding, requiring the bandwidth of 8 kbps. g729ab: Adds G.729 Annex B coding on the basis of G.729a. g729br8: Specifies G.729 Annex B coding, requiring the bandwidth of 8 kbps. This coding supports voice activity detection (VAD), discontinuous transmission (DTX), and comfort noise (CNG). g726r16: Specifies G.726 Annex A coding. It uses the adaptive differential pulse code modulation (ADPCM) technology, requiring the bandwidth of 16 kbps. g726r24: Specifies G.726 Annex A coding. It uses ADPCM, requiring the bandwidth of 24 kbps. g726r32: Specifies G.726 Annex A coding. It uses ADPCM, requiring the bandwidth of 32 kbps. g726r40: Specifies G.726 Annex A coding. It uses ADPCM, requiring the bandwidth of 40 kbps. 1-10 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the compression command to configure the voice coding method according to priority level. Use the undo compression command to restore the default value. By default, g729r8 coding mode is set. g711alaw and g711ulaw coding provide high-quality voice transmission, while requiring greater bandwidth. g726r16, g726r24, g726r32, and g726r40 are widely adopted now for voice coding. They use the ADPCM technology and provide multiple bandwidth options. g723r53 and g723r63 coding provide silence suppression technology and comfort noise, the relatively higher speed output is based on multi-pulse multi-quantitative level technology and provides relatively higher voice quality to certain extent, and the relatively lower speed output is based on the Algebraic-Code-Excited Linear-Prediction technology and provides greater flexibility for application. The voice quality provided by the g729r8 and g729a coding is similar to the ADPCM of 32 kbps, having the quality of a toll, and also featuring low bandwidth, lesser event delay and medium processing complexity, hence it has a wide field of application. clear-channel is clear channel transmission, which is a kind of data transmission technology via VOIP. The data received from BRI or PRI is sent to IP network after encapsulation without being dealt with. The following table describes the relation between codec algorithms and bandwidth. Usually, 8000 Hz is adopted to collect voice samples. The bandwidth without compression is 64 kbps, and it is compressed using the ITU-T G series codec algorithms. The table also shows the compression ratio. Table 1-2 Relation between algorithms and bandwidth Codec algorithm Bandwidth Voice quality G.711 (A-law and µ-law) 64 kbps (without compression) Best G.726 16, 24, 32, 40 kbps Good G.729 8 kbps Good G.723 r63 6.3 kbps Fair G.723 r53 5.3 kbps Fair Actual network bandwidth is related to packet assembly interval and network structure. The longer the packet assembly interval is, the closer the network bandwidth is to the media stream bandwidth. More headers consume more bandwidth. Longer packet assembly interval results in longer fixed coding latency. 1-11 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands The following tables show the relevant packet assembly parameters without IPHC compression, including packet assembly interval, bytes coded in a time unit, and network bandwidth, etc. Thus, you can choose a suitable codec algorithm according to idle and busy status of the line and network situations more conveniently. Table 1-3 G.711 algorithm (A-law and µ-law) Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP 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 64kbps, minimum packet assembly interval 10 ms. Table 1-4 G.729 algorithm Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP Network bandwidt h IP+PPP Coding latency 10 ms 10 50 40 kbps 56 44.8 kbps 10 ms 20 ms 20 60 24 kbps 66 26.4 kbps 20 ms 30 ms 30 70 18.7 kbps 76 20.3 kbps 30 ms G.729 algorithm: media stream bandwidth 8 kbps, minimum packet assembly interval 10 ms. Table 1-5 G.723 r63 algorithm Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP Network bandwidt h IP+PPP Coding latency 30 ms 24 64 16.8 kbps 70 18.4 kbps 30 ms 60 ms 48 88 11.6 kbps 94 12.3 kbps 60 ms 90 ms 72 112 9.8 kbps 118 10.3 kbps 90 ms G.723 r63 algorithm: media stream bandwidth 6.3 kbps, minimum packet assembly interval 30 ms. 1-12 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Table 1-6 G.723 r53 algorithm Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP Network bandwidt h IP+PPP Coding latency 30 ms 20 60 15.9 kbps 66 17.5 kbps 30 ms 60 ms 40 80 10.6 kbps 86 11.4 kbps 60 ms 90 ms 60 100 8.8 kbps 106 9.3 kbps 90 ms G.723 r53 algorithm: media stream bandwidth 5.3 kbps, minimum packet assembly interval 30 ms. Table 1-7 G.726 r16 algorithm Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP Network bandwidt h IP+PPP Coding latency 10 ms 20 60 48 kbps 66 52.8 kbps 10 ms 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 23.4 kbps 50 ms 60 ms 120 160 21.3 kbps 166 11.4 kbps 60 ms 70 ms 140 180 20.6 kbps 186 21.3 kbps 70 ms 80 ms 160 200 20 kbps 206 20.6 kbps 80 ms 90 ms 180 220 8.8 kbps 226 9.3 kbps 90 ms 100 ms 200 240 19.2 kbps 246 19.7 kbps 100 ms 110 ms 220 260 18.9 kbps 266 19.3 kbps 110 ms G.726 r16 algorithm: media stream bandwidth 16 kbps, minimum packet assembly interval 10 ms. Table 1-8 G.726 r24 algorithm Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP Network bandwidt h IP+PPP Coding latency 10 ms 30 70 56 kbps 76 60.8 kbps 10 ms 20 ms 60 100 40 kbps 106 42.4 kbps 20 ms 30 ms 90 130 34.7 kbps 136 17.5 kbps 30 ms 40 ms 120 160 32 kbps 166 33.2 kbps 40 ms 1-13 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP Network bandwidt h IP+PPP Coding latency 50 ms 150 190 30.4 kbps 196 31.2 kbps 50 ms 60 ms 180 220 29.3 kbps 226 11.4 kbps 60 ms 70 ms 210 250 28.6 kbps 256 30.1 kbps 70 ms G.726 r24 algorithm: media stream bandwidth 24 kbps, minimum packet assembly interval 10 ms. Table 1-9 G.726 r32 algorithm Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP Network bandwidt h IP+PPP Coding latency 10 ms 40 80 64 kbps 86 68.8 kbps 10 ms 20 ms 80 120 48 kbps 126 50.4 kbps 20 ms 30 ms 120 160 42.7 kbps 166 44.3 kbps 30 ms 40 ms 160 200 40 kbps 206 41.2 kbps 40 ms 50 ms 200 240 38.4 kbps 246 39.4 kbps 50 ms G.726 r32 algorithm: media stream bandwidth 32 kbps, minimum packet assembly interval 10 ms. Table 1-10 G.726 r40 algorithm Packet assembly interval Bytes coded in a time unit Packet length IP Network bandwidt h IP Packet length IP+PPP Network bandwidt h 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. 1-14 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Note: z Packet assembly interval is the duration to encapsulate information into a voice packet. z Bytes coded in a time unit = packet assembly interval X media stream bandwidth. z Packet length (IP) = IP header + RTP header + UDP header + voice information length = 20+12+8+data z Packet length (IP+PPP) = PPP header + IP header + RTP header + UDP header + voice information length = 6+20+12+8+data z Network bandwidth = Bandwidth of the media stream X packet length / bytes coded in a time unit Since IPHC compression is affected significantly by network stability, it cannot achieve high efficiency unless line is of high quality, 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, IP (RTP) header can be compressed to 2 bytes. If PPP header is compressed at the same time, a great deal of media stream bandwidth can be saved. The following table shows the best IPHC compression efficiency of codec algorithms with packet assembly interval of 30ms. Table 1-11 Compression efficiency of IPHC+PPP header Coding modes Bytes coded in a time unit After IPHC+PPP compression Before compression Packet length IP+PPP Network bandwidth IP+PPP Packet length IP+PPP Network bandwidth IP+PPP G.729 30 76 20.3 kbps 34 9.1 kbps G.723r63 24 70 18.4 kbps 28 7.4 kbps G.723r53 20 66 17.5 kbps 24 6.4 kbps G.726r16 60 106 28.3 kbps 64 17.1 kbps G.726r24 90 136 17.5 kbps 94 25.1 kbps G.726r32 120 166 44.3 kbps 124 33.1 kbps G.726r40 150 196 52.3 kbps 154 41.1 kbps Only when there is an intersection (a compression mode recognized by both parties) in the voice compression modes owned by the two communication parties can the two parties establish normal communication with each other. If there is no consistency in the compression modes set for the equipment at the two ends connected to each other, or there is no common compression method, the calling will fail. 1-15 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Example # Configure to select g723r53 compression method first, then to select the g729r8 compression method. [H3C-voice-dial-entity1] compression 1st-level g723r53 [H3C-voice-dial-entity1] compression 2nd-level g729r8 1.1.12 cptone Syntax cptone { locale | cs } [ { type | all } amplitude value ] undo cptone [ { locale | cs } { type | all } amplitude ] View Voice subscriber-line view Parameter Locale: Country mode, sets the prompt tone played on the current voice subscriber-line to the specified country or area mode. Now, it supports up to 62 countries or areas. cs: Custom, sets the prompt tone played on the current voice subscriber-line to the customer-defined mode. Type: Tone type, Now, it supports the following: dial-tone, special-dial-tone, busy-tone, congestion tone, ringback-tone, waiting-tone. all: All types of tone. amplitude value: Sets the tone amplitude, in the range 200 to 2000. The amplitude of busy-tone and congestion-tone defaults to 1000, that of dial tone and special dial tone defaults to 400, and that of others defaults to 600. Table 1-12 Country mode of prompt tone Type of prompt tone Description AR Argentina AU Australia AT Austria BE Belgium BR Brazil BG Bulgaria CA Canada CL Chile CN China 1-16 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Type of prompt tone Description HR Croatia CU Cuba CS Custom CY Cyprus CZ Czech Republic DK Denmark EG Egypt FI Finland FR France DE Germany GH Ghana GR Greece HK Hong Kong China HU Hungary IS Iceland IN India ID Indonesia IR Iran IE Ireland IEU Ireland UK style IL Israel IT Italy JP Japan JO Jordan KE Kenya KR Korea Republic LB Lebanon LU Luxembourg MY Malaysia MX Mexico NP Nepal NL Netherlands 1-17 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Type of prompt tone Description NZ New Zealand NG Nigeria NO Norway PK Pakistan PA Panama PH Philippines PL Poland PT Portugal RU Russian Federation SA Saudi Arabia SG Singapore SK Slovakia SI Slovenia ZA South Africa ES Spain SE Sweden CH Switzerland TH Thailand TR Turkey GB United Kingdom US United States UY Uruguay ZW Zimbabwe Description Use the cptone command to set the prompt tone played on the current voice subscriber-line to the specified country mode or customer-defined mode. Use the undo cptone command to restore the default, that is, CN. 1-18 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Note: The command will be valid for all the voice ports on the veneer of the current voice subscriber-lines. Example # Set the country mode of prompt tone to US. [H3C-voice-line1/0/0] cptone us. 1.1.13 debugging voice cm Syntax debugging voice cm { all | error | message | event | fsm | timer } undo debugging voice cm { all | error | message | event | fsm | timer } View User view Parameter all: Enables debugging on all CM. error: Enables debugging on CM errors. message: Enables debugging for the detailed information received and transmitted by CM. event: Enables debugging on the “event” information received and transmitted by CM. fsm: Enables debugging for “state” transition information in CM state machine. timer: Enables information debugging for the timer on the state machine in CM. Description Use the debugging voice cm command to enable debugging for CM. Use the undo debugging voice cm command to disable the debugging. By default, CM module debugging is disabled. Example # Enable all debugging on CM. [H3C] debugging voice cm all 1-19 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.14 debugging voice data-flow Syntax debugging voice data-flow { all | verbose | error | fax | fax-error | jitter | jitter-error | receive | send | vpp } View User view Parameter all: Enable debugging on all voice data processes. verbose: Enable debugging on detailed information of voice packets. error: Enable debugging on voice data errors. fax: Enable debugging for fax data stream. fax-error: Enable debugging on fax data errors. jitter: Enable jitter buffer debugging. jitter-error: Enable debugging for jitter buffer processing errors. receive: Enable debugging output at the receiving side of the data stream. send: Enable debugging at the sending side of the data stream. vpp: Enable debugging on the data stream of VPP software module. Description Use the debugging voice data-flow command to enable debugging for voice data processes. Use the undo debugging voice data-flow command to disable the debugging. Example # Enable debugging on all voice data processes. debugging voice data-flow all 1.1.15 debugging voice dpl Syntax debugging voice dpl { all | error | general } View User view 1-20 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter all: Enable all debugging of voice dial program. error: Enable error debugging of voice dial program. general: Enable general debugging of voice dial program. Description Use the debugging voice dpl command to enable the debugging of voice dial program. Use the undo debugging voice dpl command to disable the debugging. Example # Enable all debugging of voice dial program. debugging voice dpl all 1.1.16 debugging voice h225 Syntax debugging voice h225 { asn1 | event } View User view Parameter asn1: Information related to negotiation messages is output. event: Information related to negotiation events is output. Description Use the debugging voice h225 command to enable debugging for the H.225.0 negotiation messages or events. Use the undo debugging voice h225 command to disable the debugging. Example # Enable H.225 negotiation events debugging. debugging voice h225 event 1.1.17 debugging voice h245 Syntax debugging voice h245 | event 1-21 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View User view Parameter event: Information related to negotiation events is output. Description Use the debugging voice h245 command to enable debugging for the H.245 negotiation messages or events. Use the undo debugging voice h245 command to disable the debugging. Example # Enable H.245 negotiation events debugging. debugging voice h245 event 1.1.18 debugging voice ipp Syntax debugging voice ipp { all | error | rtp-rtcp | socket | vcc | vpp | x691 } View User view Parameter all: Enable all debugging for IPP module. error: Enable error debugging for IPP module. rtp-rtcp: Enable RTP/RTCP information debugging. socket: Enable Socket information debugging. vcc: Enable VCC message receiving and sending debugging. vpp: Enable VCC message receiving and sending debugging. x691: Enable X.691 message debugging. Description Use the debugging voice ipp command to enable H.323 recommendation suite module debugging. Use the undo debugging voice ipp command to disable this debugging. Example # Enable all debugging on IPP module. 1-22 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands debugging voice ipp all 1.1.19 debugging voice rcv Syntax debugging voice rcv { all | cc | error | r2 | timer | vas | vcc | vpp } View User view Parameter all: Enables all the debugging of the RCV module. cc: Enables the debugging between the RCV module and the underlying CC module. error: Enables the debugging of RCV-caused connection failures. r2: Displays information between the RCV module and the R2 module. timer: Enables the debugging for the timer operation of the RCV module. vas: Enables the debugging between the RCV module and the underlying VAS module. vcc: Enables the debugging between RCV and the underlying VCC module. vpp: Enables the debugging between the RCV module and the underlying VPP module. Description Use the debugging voice rcv command to enable the debugging of the RCV module. Use the undo debugging voice rcv command to disable the debugging. Example # Enable all the debugging of the RCV module. debugging voice rcv all 1.1.20 debugging voice vas Syntax debugging voice vas { all | buffer | cid | command | dsp | em | error | fax | line | rcv | receive | send } View User view 1-23 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter all: Enable all the debugging of the VAS module. buffer: Enable the debugging of the buffer area for the VAS module to transmit commands to the DSP module. cid: Enable VAS CID debugging command: Enable the debugging between the VAS module and the command buffer area. dsp: Enable the debugging between the VAS module and the underlying DSP module. em: Enable the debugging of the EM subscriber-line operation of the VAS module. error: Enable the debugging of the connection failure caused by the VAS module. fax: Enable VAS fax debugging. line: Enable the debugging for the VAS module to log on to a specified line. rcv: Enable the debugging between VAS and RCV. receive: Enable the debugging for the VAS module to receive data. send: Enable the debugging for the VAS module to transmit data. Description Use the debugging voice vas command to enable the VAS module debugging. Use the undo debugging voice vas command to disable the debugging. Example # Enable all the debugging of the VAS module. debugging voice vas all 1.1.21 debugging voice vmib Syntax debugging voice vmib { aaaclient | all | analogif | callactive | callhistory | dialcontrol | digitalif | error | general | gkclient | h323statistic | voiceif } undo debugging voice vmib { aaaclient | all | analogif | callactive | callhistory | dialcontrol | digitalif | error | general | gkclient | h323statistic | voiceif } View User view Parameter aaaclient: Enable the debugging of the AAA client. all: Enable all the debugging of the voice MIB. 1-24 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands analogif: Enable the debugging of the analog voice interfaces. callactive: Enable the debugging of the current call. callhistory: Enable the debugging of the history calls. dialcontrol: Enable the dialing debugging. error: Enable debugging for voice MIB errors. general: Enable the general voice MIB debugging. gkclient: Enable the GK client debugging. h323statistic: Enable debugging for 323 statistics. voiceif: Enable the voice interface debugging. Description Use the debugging voice vmib command to enable the voice MIB module debugging. Use the undo debugging voice vmib command to disable the debugging. Example # Enable all voice MIB debugging. debugging voice vmib all 1.1.22 debugging voice vpp Syntax debugging voice vpp { all | codecm | error | ipp | r2 | rcv | timer | vas | vcc } View User view Parameter all: Enable all the debugging of the VPP module. codecm: Enable the debugging between the VPP module and the underlying CODECM module. error: Enable the debugging of the connection failure of the VPP module. ipp: Enable the debugging between the VPP module and the upper layer IPP module. r2: Enable the debugging of information between VPP and R2 rcv: Enable the debugging between the VPP module and the RCV module timer: Enable the debugging for the timer operation of the VPP module. vas: Enable the debugging between the VPP module and the VAS module. vcc: Enable the debugging of information between VPP and VCC 1-25 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the debugging voice vpp command to enable the debugging of the VPP module. Use the undo debugging voice vpp command. you can disable the debugging switch. Example # Enable all the debugging of the VPP module. debugging voice vpp all 1.1.23 default entity compression Syntax default entity compression { 1st-level | 2nd-level | 3rd-level | 4th-level } { g711alaw | g711ulaw | g723r53 | g723r63 | g729a | g729r8 | g729ab | g729br8 | g726r16 | g726r24 | g726r32 | g726r40 } undo default entity compression { 1st-level | 2nd-level | 3rd-level | 4th-level } View Voice dial program view Parameter Refer to the compression command. Description Use the default entity compression command to globally configure the mode of coding and decoding as the default value. Use the undo default entity compression command to restore the fixed value (i.e. g729r8 code mode) in the system as the default value. By default, the mode of coding and decoding is g729r8 coding mode. The default entity compression command can be used to globally configure the default value of the voice coding and decoding. In this case, all the voice entities and newly created voice entities on this router, which have not been configured with this function will inherit this configuration. Related command: compression. Example # Adopt the g723r53 coding and decoding mode as the first selection globally. [H3C-voice-dial] default entity compression 1st-level g723r53 1-26 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.24 default entity normal-connect slow-h245 Syntax default entity normal-connect slow-h245 undo default entity normal-connect slow-h245 View Voice dial program view Parameter None Description Use the default entity normal-connect slow-h245 command to globally disable the calling end from actively initiating an H245 connection request to the called end before the called end is hooked off. Use the undo default entity normal-connect slow-h245 command to restore the default value (that is, to allow the calling end to actively initiate an H245 connection request to the called end before the called end is hooked off.). By default, the calling end is allowed to actively initiate an H245 connection request to the called end before the called end is hooked off. Use the default entity normal-connect slow-h245 command to default to globally disable the calling end from actively initiating the H245 connection request to the called end before the called end is hooked off. In this case, all newly created voice entities and voice entities which have not been configured with this function will inherit this configuration. Related command: normal-connect slow-h245. Example # Configure to globally disable the calling end from actively initiating an H245 connection request to the called end before the called end is hooked off. [H3C-voice-dial] default entity normal-connect slow-h245 1.1.25 default entity payload-size Syntax default entity payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } undo default entity payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } 1-27 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View Voice dial program view Parameter g711: Specifies the time length of voice packets with g711 coding. It can be 10 milliseconds, 20 milliseconds (the default), or 30 milliseconds. g723: Specifies the time length of voice packets with g723 coding, in the range 30 to 180 milliseconds. It defaults to 30 milliseconds. g726r16: Specifies the time length of voice packets with g726r16 coding. It ranges from 10 to 110 milliseconds and defaults to 30 milliseconds. g726r24: Specifies the time length of voice packets with g726r24 coding. It ranges from 10 to 70 milliseconds and defaults to 30 milliseconds. g726r32: Specifies the time length of voice packets with g726r32 coding. It ranges from 10 to 50 milliseconds and defaults to 30 milliseconds. g726r40: Specifies the time length of voice packets with g726r40 coding. It ranges from 10 to 40 milliseconds and defaults to 30 milliseconds. g729: Specifies the time length of voice packets with g729 coding, in the range10 to 180 milliseconds. It defaults to 30 milliseconds. Description Use the default entity payload-size command to configure the default time length of voice packets with different coding formats. Use the undo default entity payload-size command to restore the default. Example # Set the time length of voice packets with G.711 coding to 30 milliseconds. [H3C-voice-dial] default entity payload-size g711 30 1.1.26 default entity service data enable Syntax default entity service data enable undo default entity service data enable View Voice dial program view Parameter None 1-28 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the default entity service data enable command to globally configure enabling data call service. Use the undo default entity vad-on enable command to disable the global data call service. By default, the global data call service is disabled. Example # Enable the data call service globally. [H3C-voice-dial] default entity service data enable 1.1.27 default entity service data payload-size Syntax default entity service data payload-size payload-size undo default entity service data payload-size View Voice dial program view Parameter payload-size: Length of packetizing RTP data for an ISDN data call. It can be 10, 20, or 30 milliseconds. The default packetization length is 20 milliseconds. Description Use the default entity service data payload-size command to configure the length of packetizing RTP data for an ISDN data call. In ISDN data calls, data is carried in voice RTP packets. The packetization length of RTP data is as adjustable as that of VoIP RTP data is. Three payload sizes are currently supported: 80 bytes, 160 bytes, and 240 bytes. Their corresponding packetization lengths are 10, 20, and 30 milliseconds, respectively. You can configure this command globally, but cannot in voice entity view. Use the undo default entity service data payload-size command to restore the default. Example # Set the packetization length for an ISDN call to 30 milliseconds. [H3C-voice-dial]default entity service data payload-size 30 1-29 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.28 default entity vad-on Syntax default entity vad-on undo default entity vad-on View Voice dial program view Parameter None Description Use the default entity vad-on command to globally configure enabling silence detection as the default value. Use the undo default entity vad-on command to restore the fixed value (i.e. disabling the silence detection) to be the default value. By default, the silence detection is disabled. The default entity vad-on command is used to globally configure enabling silence detection as the default value. In this case, all the voice entities and newly created voice entities on this router, which have not been configured with this function, will inherit this configuration. Related command: vad-on. Example # Enable the silence detection globally. [H3C-voice-dial] default entity vad-on 1.1.29 delay Syntax delay { dtmf | dtmf-interval } milliseconds delay start-dial seconds undo delay { dtmf | dtmf-interval start-dial } View FXO voice subscriber-line view 1-30 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter dtmf milliseconds: Lasting duration of a DTMF signal on the FXO interface, in the range of 50 to 500 milliseconds with a default of 120. dtmf-interval milliseconds: Interval between two DTMF signals on the FXO interface, which is in the range of 50 to 500ms and defaults to 120ms. start-dial seconds: Delay in seconds in dialing on the FXO interface, in the range of 0 to 10 with a default of 1. Description Use the delay command to configure the relevant time parameters on an FXO or analog E&M subscriber-line. Use the undo delay command to restore the default values of these time parameters. All the commands listed above are used for configuring the device of a calling party and hence are only useful for the calling party. Related command: timer. Example # Set the hold time for delay on line 1/0/0 to 5s. [H3C-voice-line1/0/0] delay start dial 5 1.1.30 delay-reversal Syntax delay-reversal seconds undo delay-reversal View Voice subscriber line view Parameter seconds: Length of the timer of polarity-reverse transmission delay, in the range 10 to 30 seconds. Description Use the delay-reversal command to configure the timer’s interval of polarity-reverse transmission delay on FXS interfaces. Use the undo delay-reversal command to remove the function of polarity-reverse transmission delay. This command applies to FXS interfaces. 1-31 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands By default, FXS interfaces immediately send polarity-reverse signal once receiving the hookoff signal from the remote end without any delay. Example # Set the delay interval of voice subscriber line 1 to 15 seconds. [H3C-voice-line1] delay-reversal 15 1.1.31 description (voice entity view) Syntax description string undo description View Voice entity view Parameter string: Voice entity description string, with length ranging from 1 to 64 characters. Description Use the description command to configure a voice entity description string. Use the undo description command to delete the voice entity description string. You can make a description of the voice entity by using the description command. This operation will not affect the performance of voice entity interfaces at all. You can view its information when executing the display command. Example # Identify voice entity 10 with local-entity 10. [H3C-voice-dial-entity10] description local-entity10 1.1.32 description (voice subscriber line view) Syntax description string undo description View FXO voice subscriber-line view 1-32 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter string: Subscriber-line description character string, and its value range is 1 to 64 characters. Description Use the description command to configure a subscriber-line description character string. Use the undo description command to cancel the subscriber-line description of character string. This command is applicable to FXO, FXS, E1V1 interfaces. With the description command, make a description of the voice subscriber-line connection. This operation will not have any influence on the running of voice entities, only when the display command is being executed will the configuration information be seen. Example # Identify subscriber line 1/0/0 as connected to lab_1. [H3C-voice-line1/0/0] description lab_1 1.1.33 dial-prefix Syntax dial-prefix string undo dial-prefix View Voice entity view Parameter string: Prefix code, a number of fixed length. The string is composed of any characters from “0123456789 #*”, with the largest length of 31 characters. The meanings of the characters are shown in the following table. Table 1-13 Meanings of the characters in string Character Means 0-9 Numbers from 0 to 9. Each means a digit. , Pause of 500ms. It can be placed on any position of the number. # and * Valid digits. 1-33 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the dial-prefix command to configure the prefix of the telephone number dialed by the voice entity. Use the undo dial-prefix command to cancel the prefix of the telephone number dialed by the voice entity. This command only applies to the configuration of POTS voice entity. And the dial-prefix command only applies to FXO, BSV and analog E&M interface. Whether to send a second stage dialing tone is determined by the configuration of the PBX connected to the router. When a router with which a voice function is configured receives a voice call, it makes a comparison between the number configured in the match-template of its own POTS voice entity and the number received, and selects one POTS voice entity to continue the call processing. If send-number is set to its default value (truncate), the router will remove from the called number the string that matches the match-template beginning from the left. If the dial-prefix command is configured, the prefix will be added in front of the rest of the called number. The router will initiate a call according to the new number string. For example, supposing that the called number is 0102222, the called number template of the voice entity that is configured by match-template is 010…., and the dial prefix is 0, then “010” that accurately matches the template will be removed, and the rest part “2222” will be added a prefix “0”. The router initiates a call to the new number 02222. If the number with the added prefix contains more than 31 digits, only the first 31 characters will be sent. Related command: match-template and send-number. The router will remove the string of the called number that matches the match-template starting from the left. If a dial-prefix is configured, the prefix will be added before the remaining number string, and the router will make a call according to the new string generated. Suppose the called number is 0102222, the 010 on the left will be deleted because of matching with the match-template. The remaining 2222 will be added with a prefix “0” and will be dialed out again, that is, the number called by the router is 02222. This command is used only for the POTS voice entity configuration. It is valid only for FXO and E&M subscriber-lines. Related command: match-template. Example # Use 0 as a prefix. [H3C-voice-dial-entity3] dial-prefix 0 1-34 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.34 dial-program Syntax dial-program View Voice view Parameter None Description Use the dial-program command to enter the voice dial program view. Use the quit command to return to the voice view. Example # Enter the dial program view. [H3C-voice] dial-program 1.1.35 display voice call-history-record Syntax display voice call-history-record { callednumber number | callingnumber number | cardnumber number | last number | line number | remote-ip-addr ip-address } [ brief ] View Any view. Parameter callednumber number: Displays the call history of the specified called number. callingnumber number: Displays the call history of the specified calling number. cardnumber number: Displays the call history of the specified card number. last number: Displays the last n calls, with n in the range of 1 to 500. remote-ip-addr ip-address: Displays the call history of the remote IP address. brief: Displays information briefly. Description Use the display voice call-history-record command to view call history. 1-35 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Note: The specified called number cannot be greater than the actual voice subscriber-line number; otherwise, the input is invalid. Example # Display information about the calls on the specified voice subscriber-line. [H3C] display voice call-history-record line 0 Subscriber-line 0 type FXS POTS , Line state is opened start outgoing call 72 times, 48 success receive incoming call 9 times, 6 success the latest 10 called number is: %1% called number 900 %2% called number 900 %3% called number 900 %4% called number 900 %5% called number 17912 %6% called number 17920 %7% called number 17920 %8% called number 17920 %9% called number 2186 %10% called number 2526 Table 1-14 Description on the fields of display voice call-history-record Field Description Subscriber-line Index number of the subscriber-line type Interface type of the subscriber-line Line state Line status of the subscriber-line start outgoing call Total number of outgoing calls and number of successful outgoing calls on the subscriber-line receive incoming call Total number of incoming calls and number of successful incoming calls on the subscriber-line the latest 10 called number The most recent ten called numbers out the subscriber-line 1-36 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.36 display voice call-info Syntax display voice call-info { brief | verbose | mark tag } View Any view Parameter brief: Display the call information table in brief. verbose: Display the call information table in detail. mark tag: Display the call information table by tag (in the range 0 to 127). Description Use the display voice call-info command to view the call information table, including: channel number of the call, reference counter of all voice modules, module ID in use, list of the voice entities that can be selected by the current call, and the voice entity used by the current call. Example # Display the call information table of a certain time in brief: [H3C] display voice call-info brief The information table for current calls in brief # # CALL ( 0): Channel <--> 0 Module ID <--> RCV VAS VPP End # Display the call information table of a certain time in detail: [H3C] display voice call-info verbose The information table for current calls in detail # **************** CALL 0 *************** Channel number : 0 Reference counter : 4 Module check ID RCV VCC IPP : VPP Current used voice entity: 180 Voice entities are offered: 196 132 124 284 188 180 100 1-37 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands # **************** CALL 1 *************** Channel number : 1 Reference counter : 5 Module check ID RCV VAS VCC : IPP VPP Current used voice entity : 196 Voice entities are offered : 196 132 124 284 # End 1.1.37 display voice default Syntax display voice default all View Any view Parameter None Description Use the display voice default command to view the current default values and the system-fixed default values for voice and fax. For example, truncated called number is used according to the default settings and system-fixed default settings. For example, the carrier transmission energy level of GW defaults to 10 (the system-fixed default value is 15). Example # Display the current default values and the system-default default values. [H3C] display voice default all default entity fax ecm off(system: off) default entity fax protocol t38(system: t38) default entity fax protocol t38 hb-redundancy 0(system: 0) default entity fax protocol t38 lb-redundancy 0(system: 0) default entity fax level 15(system: 15) default entity fax local-train threshold 10(system: 10) default entity fax baudrate voice(system: voice) default entity fax nsf-on off(system: off) 1-38 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands default entity fax support-mode rtp(system: rtp) default entity fax train-mode ppp(system: ppp) default entity compression 1st-level g729r8(system: g729r8) default entity compression 2nd-level g711alaw(system: g711alaw) default entity compression 3rd-level g711ulaw(system: g711ulaw) default entity compression 4th-level g723r53(system: g723r53) default entity vad-on off(system: off) default entity payload-size g711 20(system: 20) default entity payload-size g723 30(system: 30) default entity payload-size g726r16 30(system: 30) default entity payload-size g726r24 30(system: 30) default entity payload-size g726r32 30(system: 30) default entity payload-size g726r40 30(system: 30) default entity payload-size g729 30(system: 30) Table 1-15 Description on the fields of the display voice default command Field Description fax ecm ECM mode is used for Fax fax protocol Fax protocol for intercommunication fax redundancy hb-redundancy Number of high-speed redundant packets, available for Fax protocol H.323-T.38 or T.38 fax redundancy lb-redundancy Number of low-speed redundant packets, available for Fax protocol H.323-T.38 or T.38 fax level Gateway carrier transmitting energy level fax local-train threshold Fax local training threshold percentage fax baudrate Highest Fax rate fax nsf-on Fax capacity negotiation mode fax support-mode Fax transmission format fax train-mode Fax training mode compression 1st-level Voice coding mode of the first priority compression 2nd-level Voice coding mode of the second priority compression 3rd-level Voice coding mode of the third priority compression 4rd-level Voice coding mode of the fourth priority cancel-truncate Truncation of called number canceled vad-on Voice entity VAD fast connect Fast connection of the calling VoIP voice entity payload-size g711 Voice entity packet assembly interval (G.711) 1-39 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Field Description payload-size g723 Voice entity packet assembly interval (G.723) payload-size g729 Voice entity packet assembly interval (G.729) 1.1.38 display voice entity Syntax display voice entity { all | pots | voip |vofr| mark entity-tag } View Any view Parameter all: All voice entities. pots: All POTS voice entities. voip: All VoIP voice entities. vofr: All VoFR voice entities. mark: Displays a voice entity. entity-tag: Tag of the voice entity that is to be displayed, ranging from 1 to 2147483647. Description Use the display voice entity command to view the configuration information of voice entities of different types. Usually, you can view the information of all the interfaces that are active in the router and the global configuration by executing the display current-configuration command. But it will display a great deal of information. So if you just want to view the configuration information of voice entities, you can use the display voice entity command. Example # Display the configuration information of POTS voice entities. [H3C] display voice entity pots Current configuration of entities ! entity 66 pots match-template 6600.. shutdown compression 1st-level g711alaw 1-40 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands dial-prefix 6600 line 6 ! entity 67 pots match-template 6600.. shutdown compression 1st-level g711alaw dial-prefix 6600 line 7 ! End 1.1.39 display voice ipp Syntax display voice ipp ccb [channel channel-number] display voice ipp statistic { all | h225 | h245 | ras | socket | timer | vcc | vpp } View Any view Parameter ccb: Displays information about the call control block in the IPP module. channel: Displays information of a voice channel in the IPP module. statistic: Displays statistics about the IPP module. all: Displays all statistics about the IPP module. 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. vcc: Displays statistics about messages between the IPP module and the VCC module. vpp: Displays statistics about messages between the IPP module and the VPP module. Description Use the display voice ipp command to view statistics about the IPP module. 1-41 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Example # Display statistics about H.225 messages of the IPP module. [VG] display voice ipp statistic h225 statistics about H225 : { Send_Setup : 0 Send_CallProceeding : 0 Send_Alerting : 0 Send_Connect : 0 Send_ReleaseComplete : 0 Send_FacilityIndUserInput : 0 Send_FacilityTCSRequest : 0 Send_FacilityTCSAck : 0 Send_FacilityTCSReject : 0 Send_FacilityOLCRequest : 0 Send_FacilityOLCAck : 0 Send_FacilityOLCReject : 0 Send_FacilityMSDRequest : 0 Send_FacilityMSDAck : 0 Send_FacilityMSDReject : 0 Send_FacilityCLCRequest : 0 Send_FacilityCLCAck : 0 Send_FacilityStartH245 : 0 Send_Error : 0 Recv_Setup : 0 Recv_CallProceeding : 0 Recv_Alerting : 0 Recv_Connect : 0 Recv_ReleaseComplete : 0 Recv_Progress : 0 Recv_FacilityTCSRequest : 0 Recv_FacilityTCSAck : 0 Recv_FacilityTCSReject : 0 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_Unknow : 0 1-42 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands } 1.1.40 display voice number-substitute Syntax display voice number-substitute [ list-tag ] View Any view Parameter list-tag: Serial number of the number substitution rule list, ranging from 1 to 2147483647. Description Use the display voice number-substitute command to view the configuration information of number substitution rule lists. It can display the information of a certain list and all the lists. Related command: number-substitute. Example # Display all the configured number substitution rule lists. [H3C] display voice number-substitute Current configuration of number substitute ! ************ NUMBER-SUBSTITUTE ************ List-tag : 1 First-rule : INDEX_INVALID Dot-match : end-only rule 0 Input-format : ^011408 Output-format : 1408 ! End 1.1.41 display voice rcv ccb Syntax display voice rcv ccb View Any view 1-43 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter None Description Use the display voice rcv ccb command to view the information related to the call control block in the RCV module. This command is used to display the information related to the incoming and outgoing call control block, the connection status of modules, the caller status, the caller numbers and the called numbers, etc. Example # Display the information related to the call control block in the RCV module. [H3C] display voice rcv ccb RCV : CCB [ 1 ] { CallID : 0x0043 CallState : TALK VCCID : 0x004f VCCpState CCID : IPPS_CONNECTED : 0xffff CCState : CCS_CONNECTED VasID : 0xffff CallType : OUTGOING CallAttribute : 0x00000000 CallSignaling : 0x00000002 EncodeType : 0x0000001f E1Slot : 0xffffffff E1Port : 0xffffffff TimeSlot : 0xffffffff ChannelID : 0x00000003 VpuState : VS_CONNECTED VccTimer : 0x00000000 CcTimer : 0x00000000 VpuTimer : 0x00000000 CcChanMsg : 0x00000000 E1ChanMsg : 0x00000000 CallerNumber : 111 CalledNumber : 660010 prev : 0x00000000 next : 0x01409500 } 1-44 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands RCV : CCB [ 2 ] { CallID : 0x0042 CallState : TALK IppID : 0x004e IppState : IPPS_CONNECTED CcID : 0xffff CcState : CCS_CONNECTED VasID : 0x0039 CallType : INCOMING CallAttribute : 0x00000000 CallSignaling : 0x00000004 EncodeType : 0x00000009 E1Slot : 0xffffffff E1Port : 0xffffffff TimeSlot : 0xffffffff ChannelID : 0x00000000 VpuState : VS_CONNECTED IppTimer : 0x00000000 CcTimer : 0x00000000 VpuTimer : 0x00000000 CcChanMsg : 0x00000000 E1ChanMsg : 0x00000000 CallerNumber : 111 CalledNumber : 660010 prev : 0x01409000 next : 0x00000000 } Table 1-16 Description on call control block in RCV module Field Description CCB [ ] Index of call control block CallID Flag or Identifier of the calling VCCID Index of the VCC software module control block for the calling VCCState Status of the VCC software module control block for the calling CCID Index of the CC software module control block for the calling CCState Status of the CC software module control block for the calling VASID Index of the VAS software module control block for the calling 1-45 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Field Description CallType Type of the calling CallAttribute Attribute of the calling CallSignaling Signaling of the calling EncodeType Voice compression method of the calling E1Slot Number of slot where E1V1 board is located for the calling E1Port Number of CE1/PRI interface for the calling TimeSlot Timeslot on the E1 of the calling ChannelID Identifier of the logic channel of the calling VPUState Status of the VPU of the calling VCCTimer Timer of the VCC module in calling period CCTimer Timer of the CC module in calling period VPUTimer Timer of the VPU module in calling period CcChanMsg Pointer of CC channel message in calling period E1ChanMsg Pointer of E1VI channel message in calling period CallerNumber Calling number of the calling CalledNumber Called number of the calling prev Previous RCV call control block next Next RCV call control block 1.1.42 display voice rcv statistic Syntax display voice rcv statistic { all | call | cc | error | proc | r2 | timer | vas | vcc | vpp } View Any view Parameter all: Display all the statistics information of the RCV module. call: Display the calling statistics information in the RCV module. cc: Display all the statistic information between the RCV module and the underlying CC module. error: Display all the statistic information on the failure of connection caused by the RCV module. 1-46 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands proc: Display the statistic information on the number of times of the execution of various functions in the RCV module. r2: Display the statistics information between the RCV module and the R2 module. timer: Display all the statistic information on the timer operation in the RCV module. vas: Display all the statistic information between the RCV module and the underlying VAS module. vcc: Display the calling statistics information in the VCC module. vpp: Display all the statistic information between the RCV module and the underlying VPP module. Description Use the display voice rcv statistic command to view the statistics information of calling between the RCV module and the CC module, VAS module, and so on. Example # Display the statistics information of calling between the RCV module and other modules. [H3C] display voice rcv statistic call Statistic about RCV calls : { RCV_CC_ACTIVE_CALL : 0 RCV_CC_ACTIVE_CALL_SUCCEEDED : 0 RCV_CC_ACTIVE_CALL_FAILED : 0 RCV_CC_PASSIVE_CALL : 0 RCV_CC_PASSIVE_CALL_SUCCEEDED : 0 RCV_CC_PASSIVE_CALL_FAILED : 0 RCV_R2_ACTIVE_CALL : 5 RCV_R2_ACTIVE_CALL_SUCCEEDED : 2 RCV_R2_ACTIVE_CALL_FAILED : 3 RCV_R2_PASSIVE_CALL : 4 RCV_R2_PASSIVE_CALL_SUCCEEDED : 1 RCV_R2_PASSIVE_CALL_FAILED : 3 RCV_VAS_ACTIVE_CALL : 39 RCV_VAS_ACTIVE_CALL_SUCCEEDED : 11 RCV_VAS_ACTIVE_CALL_FAILED : 28 RCV_VAS_PASSIVE_CALL : 18 RCV_VAS_PASSIVE_CALL_SUCCEEDED : 15 RCV_VAS_PASSIVE_CALL_FAILED : 3 } 1-47 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Table 1-17 Description on call statistics about RCV module and other module Field Description RCV_CC_ACTIVE_CALL Number of active call between self and CC module RCV_CC_ACTIVE_CALL_SUCCEEDE D Successful number of active between self and CC module RCV_CC_ACTIVE_CALL_FAILED Failing number of active call between self and CC module RCV_CC_PASSIVE_CALL Number of passive call between self and CC module RCV_CC_PASSIVE_CALL_SUCCEED ED Successful number of passive call between self and CC module RCV_CC_PASSIVE_CALL_FAILED Failing number of passive call between self and CC module RCV_R2_ACTIVE_CALL Number of active call between self and R2 module RCV_R2_ACTIVE_CALL_SUCCEEDE D Successful number of active between self and R2 module RCV_R2_ACTIVE_CALL_FAILED Failing number of active call between self and R2 module RCV_R2_PASSIVE_CALL Number of passive call between self and R2 module RCV_R2_PASSIVE_CALL_SUCCEED ED Successful number of passive call between self and R2 module RCV_R2_PASSIVE_CALL_FAILED Failing number of passive call between self and R2 module RCV_VAS_ACTIVE_CALL Number of active call between self and VAS module RCV_VAS_ACTIVE_CALL_SUCCEED ED Successful number of active between self and VAS module RCV_VAS_ACTIVE_CALL_FAILED Failing number of active call between self and VAS module RCV_VAS_PASSIVE_CALL Number of passive call between self and VAS module RCV_VAS_PASSIVE_CALL_SUCCEE DED Successful number of passive call between self and VAS module RCV_VAS_PASSIVE_CALL_FAILED Failing number of passive call between self and VAS module 1-48 call call call Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.43 display voice sip register-state Syntax display voice sip register-state View Any view Parameter None Description Use the display voice sip register-state command to view the registration state about the SIP user agents (UAs). Example # Display the registration state about the SIP UAs. display voice sip register-state [ SIP-REG Information ] +-------------------------------------------------+ [RegFlag] -> SIP_REGISTER_FAILURE [IPAddr] -> 192.168.80.50 [Port] -> 5060 Current SIP Server IPAddr = [192.168.80.50], Port = [5060] 1.1.44 display voice subscriber-line Syntax display voice subscriber-line line-number View Any view Parameter line-number: Subscriber line number. 1-49 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the display voice subscriber-line command to view the configuration information about the type, status, codec mode, input gain and output attenuation of the subscriber line. Related command: subscriber-line. Example # Display the configuration information about the subscriber line. [H3C] display voice subscriber-line 0 Current information --- line: 0 Type = LINE FXO Status = OPEN -- CH_IDLE Coding = Decoding = CallerNum = CalledNum = Call-ID = 0 Call-Refer = 0 CNG = ON EchoCancel = ON - 32 (ms) Reset = 0 Position = Slot 2 Port 0 CID-Display = ENABLE CID-Receive = ENABLE Gain(R&T) = 0 (db) - 0 (db) T_FirstDial = 10 (s) T_DialInter = 10 (s) T_RingBack = 60 (s) T_WaitDigit = 5 (s) T_Predial = 1 (s) T_DTMF = 120 (ms) T_Interdigit= 120 (ms) Table 1-18 Description on voice subscriber-line configuration Field Description line Index of voice subscriber-line type Type of voice subscriber-line state Status of voice subscriber-line Status information Calling status of voice subscriber-line 1-50 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Field Description Use as coding protocol Voice compression subscriber-line use as decoding protocol Voice decompression subscriber-line Current ANI Calling number of current call Current DNIS Called number of current call Current direction Direction of current call Current call-ID Identifier of current call Current call-reference Call reference of current call Comfort-noise Comfort noise configuration on the voice subscriber-line reset times Reset times of the board subscriber-line is located echo cancel delay Time of echo cancel on the voice subscriber-line FXO Slot Slot number of the voice subscriber-line FXO Port Port number of the voice subscriber-line cid receive function Receive calling numbers about CID function cid display function Display calling number about CID function Current line State Status of voice subscriber-line( UP or DOWN) Receive Gain value Input gain of the voice subscriber-line Transmit Gain value Output gain of the voice subscriber-line Echo Cancellation Echo cancel function of the voice subscriber-line First-dial timer Timeout of dialing the first number on local subscriber-line Dial-interval Timer Interval timeout of dialing on local subscriber-line Ringing Time Out Timeout of sending ringing back signal on subscriber-line Wait-digit Time Out Timeout of wait digit dialed by user for subscriber-line Pre-dial delay time Delay time of pre-dial on subscriber-line DTMF digit duration Duration time subscriber-line Interdigit duration Interval time of DTMF number on subscriber-line 1-51 of method of voice of voice method DTMF where number voice on Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.45 display voice voip data-statistic Syntax display voice voip data-statistic { brief | channel channel-number ] | verbose } View Any view Parameter channel channel-number: Display statistics information of voice data on specified channel, channel-number is the logical channel number of voice, which accumulates from 0. The channel for 2VI and 4VI modularized card is numbered first and then E1V1 modularized card. verbose: Display detailed information of logical channel of voice. Description Use the display voice voip data-statistic command to view statistics information of voice data. This command is used to display the following information: time for successfully searching the voice table, total number of received data packets, time for searching the table in fast and common modes and voice and fax of receive and transmit channels etc. Used with the verbose keyword, this command displays detailed information of receive and send channels (including detailed input/output statistics and jitter Buffer of information packets). Related command: vqa data-statistic and reset voice voip data-statistic. Example # Display the statistics information of voice data. [H3C] display voice voip data-statistic brief === VoIP datagram summary === -------------------------------------------------------------------[NET] SearchVoiceTableSuccess : 0 [NET] ReceiveDatagramTotal : 0 [NET] FastSearchTableTimes : 0 [NET] NormalSearchTableTimes : 0 ------------------------ Receive channel: 000 ---------------------[NET] ReceiveDatagramTotal : 0 [COM] DiscardDatagramTotal : 0 [NET] ReceiveRtpDatagram : 0 [NET] ReceiveRtcpDatagram : 0 [COM] AddReceiveTable : 3 1-52 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands [COM] ClearReceiveTable : 0 [COM] FreeReceiveBuffer : 0 [COM] TrsmitDone : 462 VOICE DATA INFORMATION: [NET] ReceiveDataTotal : 0 [NET] NormalProcessData : 0 FAX DATA INFORMATION: [NET] ReceiveDataTotal : 0 [NET] NormalProcessData : 0 ------------------------ Send channel: 000 ---------------------- COMMON INFORMATION: [VPP] LinkReceiveDataTotal : 484 [VPP] LinkReceiveEmptyData : 0 [COM] AddTableTimes : 0 [COM] ClearTableTimes : 0 VOICE DATA INFORMATION: [FSD] FSSend_ReceiveDataTotal : 483 [COM] DiscardDataTotal : 0 [NET] NormalSendDataTotal : 0 [NET] SendToIPDataTotal : 0 [LOC] SendToLocalDataTotal : 484 FAX DATA INFORMATION: [FSD] FSSend_RcvFaxDataTotal : 0 [FDF] DiscardDataTotal : 0 [NET] SendToIPDataTotal : 0 [LOC] SendToLocalDataTotal : 0 Table 1-19 Description on statistics information of voice data Field Description SearchVoiceTableSuccess Successful times of searching voice table ReceiveDatagramTotal(summary) Number of total packets received FastSearchTableTimes Times of fast searching voice table NormalSearchTableTimes Times of normal searching voice table ReceiveDatagramTotal(channel) Number of total packets received in channel DiscardDatagramTotal Number of total packets dropped in channel ReceiveRtpDatagram Number of total RTP packets received in channel ReceiveRtcpDatagram Number of total RTCP packets received in channel AddReceiveTable Times of adding receiving table 1-53 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Field Description ClearReceiveTable Times of resetting receiving table FreeReceiveBuffer Times of clearing buffer on board TrsmitDone Times of freeing transmit buffer ReceiveDataTotal(voice) Number of total data received in channel NormalProcessData(voice) Number of total data processed normally in channel ReceiveDataTotal(fax) Number of total fax data received in channel NormalProcessData(fax) Number of total fax data processed normally in channel LinkReceiveDataTotal Number of total local data received in channel LinkReceiveEmptyData Number of total null local data received in channel AddTableTimes Times of adding sending table ClearTableTimes Times of resetting sending table FSSend_ReceiveDataTotal(voice) Number of total data fast received or sent DiscardDataTotal(voice) Number of total data dropped NormalSendDataTotal Number of total data sent normally SendToIPDataTotal(voice) Number of total data sent to IP side SendToLocalDataTotal(voice) Number of total data sent to local side FSSend_ReceiveDataTotal(fax) Number of total fax data fast received or sent DiscardDataTotal(fax) Number of total fax data dropped SendToIPDataTotal(fax) Number of total fax data sent to IP side SendToLocalDataTotal(fax) Number of total fax data sent to local side 1.1.46 display voice vpp Syntax display voice vpp [ channel channel-number ] View Any view Parameter channel-number: Voice channel number. 1-54 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the display voice vpp command to view all the statistic information in the VPP module. This command is used to display the number of times of correct and incorrect connection, the number of times of correct and incorrect disconnection, the number of times of correct and incorrect code data receiving, the number of times of correct and incorrect code data transmission, the number of times of correct and incorrect IPP data receiving, the number of times of correct and incorrect IPP data transmission, the total number of bytes of code data received, and the total number of bytes of IPP data received in various voice channels. Example # Display all statistics about channel 0 in the VPP module. [H3C] display voice vpp channel 0 Channel = 0 Status = CH_IDLE ConnectRightTimes = 0 ConnectWrongTimes = 0 DisConnectRightTimes = 0 DisConnectWrongTimes = 0 RecvCodecmDataRightTimes = 0 RecvCodecmDataWrongTimes = 0 SendCodecmDataRightTimes = 0 RecvIppDataRightTimes = 0 RecvIppDataWrongTimes = 0 SendIppDataRightTimes = 0 SendIppDataWrongTimes = 0 RecvCodecmDataBytes = 0 RecvIppDataBytes = 0 TimeJitterLess10msTimes = 0 TimeJitterLess20msTimes = 0 TimeJitterLess30msTimes = 0 TimeJitterLess40msTimes = 0 TimeJitterLess50msTimes = 0 TimeJitterLess60msTimes = 0 TimeJitterLess70msTimes = 0 TimeJitterLess80msTimes = 0 TimeJitterLess90msTimes = 0 TimeJitterLess100msTimes = 0 TimeJitterLess110msTimes = 0 RecvIppDataSeqHopeTimes = 0 RecvIppDataDisorderTimes = 0 RecvIppDataRecvSeqLessTimes = 0 RecvIppDataSeqMoreTimes = 0 ulSendNoBDTimes ulRecvExpirePacketTimes = 0 = 0 ulRecvDuplicatePacketTimes = 0 ulJitterBufferOverFlowTimes1 = 0 ulJitterBufferOverFlowTimes2= 0 ulEmptyPacketTimes ulSendDSPVOIPPacket ulSendDSPFOIPPacket ulCorputTimes = 0 = 0 1-55 = 0 = 0 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Table 1-20 Description of statistics in VPP module Field Description ConnectRightTimes Times of connecting correctly ConnectWrongTimes Times of connecting wrongly DisConnectRightTimes Times of disconnecting correctly DisConnectWrongTimes Times of disconnecting wrongly RecvCodecmDataRightTimes Times of receiving compression/decompression data correctly RecvCodecmDataWrongTimes Times of receiving compression/decompression data wrongly SendCodecmDataRightTimes Times of sending compression/decompression data correctly RecvIppDataRightTimes Times of receiving IPP data correctly RecvIppDataWrongTimes Times of receiving IPP data wrongly SendIppDataRightTimes Times of sending IPP data correctly SendIppDataWrongTimes Times of sending IPP data wrongly RecvCodecmDataBytes Number of bytes of receiving compression/decompression data RecvIppDataBytes Number of data bytes received from the IPP module TimeJitterLess(10-110)msTimes Statistics of Jitter time of receiving IPP data RecvIppDataSeqHopeTimes Times of receiving IPP data whose sequence is accordant with expectation RecvIppDataDisorderTimes Times of receiving IPP data which is out of order RecvIppDataRecvSeqLessTimes Times of receiving IPP data whose sequence is less than expectation RecvIppDataSeqMoreTimes Times of receiving IPP data whose sequence is larger than expectation ulSendNoBDTimes Times of sending data without buffer ulRecvExpirePacketTimes Times of receiving expired packets ulRecvDuplicatePacketTimes Times of receiving duplicate packets ulJitterBufferOverFlowTimes1 Times 1 of JitterBuffer overflows ulJitterBufferOverFlowTimes2 Times 2 of JitterBuffer overflows ulEmptyPacketTimes Times of empty packets ulSendDSPVOIPPacket Number of voice packets sending to DSP ulSendDSPFOIPPacket Number of fax packets sending to DSP 1-56 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Field Description ulCorputTimes Times of collision in JitterBuffer 1.1.47 dot-match Syntax dot-match { end-only | left-right | right-left } undo dot-match View Voice number-substitute view Parameter end-only: Reserve the digits that correspond to all the dots “.” at the end of the number input format. left-right: Reserve from left to right the digits that correspond to the dots in the number input format. right-left: Reserve from right to left the digits that correspond to the dots in the number input format. Description Use the dot-match command to configure dot match rules of the number substitution rules. Use the undo dot-match command to restore the default value. The configuration of this command only applies to the rules of the number substitution rule list in the current view. By default, the dot match rule is set to end-only. According to the configuration of the dot-match command, the dots can be reserved by quantity and position of the dots configured in output format of the number substitution rule. There are three dot match modes in the number substitution rules. z Reserve the digits that correspond to all the dots at the end of the number input format. It is the default mode. z Reserve from left to right the digits that correspond to the dots in the number input format. z Reserve from right to left the digits that correspond to the dots in the number input format. The dots “.” here are virtual match digits. Virtual match digits are the digits that match the variable part (such as + % ! []) in an expression. For example, when 1255 is 1-57 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands matched with 1[234]55, the virtual match digit is 2; when it is matched with 125+, the virtual match digits are 5; when it is matched with 1..5, the virtual match digits are 25. Note: For details about dot match rules in number substitution rules, refer to the rule command. Example # Set the dot match rule of the number substitution rule list 20 to right-left. [H3C-voice-dial] number-substitute 20 [H3C-voice-dial-substitute20] dot-match right-left 1.1.48 dscp media Syntax dscp media dscp-value undo dscp media View Voice entity view Parameter dscp-value: DSCP value in the range 0 to 63 or the keyword ef, af11, af12, af13, af21, af22, af23, af31, af32, af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, ef, or zero. Description Use the dscp media command to set the DSCP value in the ToS field in the IP packets that carry the RTP stream of the voice entity. Use the undo dscp media command to restore the default DSCP. The function of this command is the same as that of the vqa dscp command. Example # Set the DSCP value in the ToS field in the IP packets that carry the RTP stream of VoIP voice entity 2 to af41. [H3C-voice-dial] entity 2 voip [H3C-voice-dial-entity2] dscp media af41 1-58 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.49 dtmf sensitivity-level Syntax dtmf sensitivity-level { high | low } undo dtmf sensitivity-level View FXS/FXO voice subscriber line view Parameter high: Sets the DTMF code detection sensitivity level as high. In this mode, the reliability is lower, and some codes may be mistaken for DTMF codes. low: Sets the DTMF code detection sensitivity level as low. In this mode, the reliability is higher, but DTMF code may be missed. Description Use the dtmf sensitivity-level command to set the detection sensitivity level of DTMF codes. Use the undo dtmf sensitivity-level command to restore the default detection sensitivity level. By default, the detection sensitivity level of DTMF codes is high. This command is only valid for FXS/FXO interface. Example # Set the DTMF code detection sensitivity level of voice subscriber line 1/0 as low. system-view [H3C] voice-setup [H3C-voice] subscirber-line1/0 [H3C-voice-line1/0] dtmf sensitivity-level low 1.1.50 dtmf threshold Syntax dtmf threshold { analogue index value1 | digital value2 } undo dtmf threshold { analogue | digital } index View Voice subscriber-line view 1-59 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter analogue: Sets an analog voice subscriber-line. digital: Sets a digital voice subscriber-line. index: Index number corresponding to the threshold, an integer in the range 0 to 12. value1: Specifies the threshold corresponding to the specified index. value2: Takes 0 or 1 to set DTMF digit detection to insensitivity or sensitivity. On an E1 voice subscriber-line, DTMF digit detection defaults to 1. The following table provides the description on the index numbers. Table 1-21 Description on the index numbers Index numbers 0 1 Indicates The lower threshold of ROWMAX plus COLMAX. The input signal which is otherwise regarded too weak is recognized as a DTMF digit when ROWMAX + COLMAX) > 0. The threshold is an integer in the range 0 to 5000 and defaults to 1400. A higher threshold means increased detection reliability but decreased sensitivity. The upper threshold of the maximum value of ROWMAX or COLMAX, whichever is larger. This threshold is used for detecting the inter-digit delay. A detected digit is regarded ended only when max (ROWMAX, COLMAX) < 1. The threshold is an integer in the range 0 to 5000 and defaults to 458. A smaller value means increased detection reliability but decreased sensitivity. 2 The lower threshold of COLMAX / ROWMAX, where ROWMAX < COLMAX. A DTMF digit is regarded ideal when COLMAX ≈ ROWMAX, because the difference between the two values is small then. An input signal is recognized as a DTMF digit only when (COLMAX / ROWMAX) > 2. The threshold is an integer in the range -18 to -3 dB and defaults to -9 dB. A higher value means increased detection reliability but decreased sensitivity. 3 The lower threshold of ROWMAX / COLMAX when COLMAX >= ROWMAX. The function of this parameter is similar to that of index 2, except that they are in reverse. The ratio must be greater than this threshold for the input signal to be recognized as a DTMF digit. The threshold is an integer in the range -18 to -3 dB and defaults to -9 dB. A lower value means increased detection reliability but decreased sensitivity. 1-60 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Index numbers 4 Indicates The upper threshold of the ratio of the second largest energy level from the row frequency group to ROWMAX. The ratio must be lower than this threshold for the input signal to be recognized as a DTMF digit. The threshold is an integer in the range -18 to -3 dB and defaults to -9 dB. A smaller value means increased detection reliability but decreased sensitivity. 5 The upper threshold of the ratio of the second largest energy level from the column frequency group to COLMAX. The ratio must be lower than this threshold for the input signal to be recognized as a DTMF digit. The threshold is an integer in the range -18 to -3 dB and defaults to -9 dB. A smaller value means increased detection reliability but decreased sensitivity. 6 7 8 The upper threshold of ROW2nd / ROWMAX. An input signal is recognized as a DTMF digit only when ROW2nd / ROWMAX < 6. The threshold is an integer in the range -18 to -3 dB and defaults to -3 dB. A smaller value means increased detection reliability but decreased sensitivity. The upper threshold of COL2nd / COLMAX. The ratio must be lower than this threshold for the input signal to be recognized as a DTMF digit. The threshold is an integer in the range -18 to -3 dB and defaults to -12 dB. A smaller value means increased detection reliability but decreased sensitivity. The upper threshold of the ratio of the maximum value of the larger energy level among two extra specified frequency points to max (ROWMAX, COLMAX). The ratio must be greater than this threshold for the input signal to be recognized as a DTMF digit. The threshold is an integer in the range -18 to -3 dB and defaults to -12 dB. A smaller value means increased detection reliability but decreased sensitivity. 9 10 The lower threshold of the DTMF signal duration. The duration of DTMF key tone must be larger than this threshold for the input signal to be recognized as a DTMF digit. The threshold is in the range 30 to 150 milliseconds and defaults to 30 milliseconds. A greater value means increased detection reliability but decreased sensitivity. The frequency of the first extra frequency point specified for detection. It is an integer in the range 300 to 3400 Hz and defaults to 300 Hz. In addition, it must be a value beyond the row and column frequency groups. 1-61 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Index numbers 11 Indicates The frequency of the second extra frequency point specified for detection. It is an integer in the range 300 to 3400 Hz and defaults to 3200 Hz. In addition, it must be a value beyond the row and column frequency groups. The lower threshold of the amplitude of the input signal. The average amplitude must be greater than this threshold for the input signal to be recognized as a DTMF digit. 12 This threshold is an integer in the range 0 to 700 and defaults to 375. A greater value means increased detection reliability but decreased sensitivity. This parameter is a time domain threshold, which is specified to prevent the noise with small amplitude from being detected. Description Use the dtmf threshold command to configure the sensitivity of DTMF digit detection. Use the undo dtmf threshold command to restore the default. The dtmf threshold command issues the thresholds for DTMF dial tone detection to the underlying layer DSP for the purpose of tuning detection sensitivity and reliability of the device subtly. Inside the DSP, a set of generic default values have been configured. They are 1400, 458, -9, -9, -9, -9, -3, -12, -12, 30, 300, 3200, 375, with their index being 0 through 12. Professionals can use this command to adjust the device when DTMF digit detection fails. In normal cases, the defaults are adequate. When the value2 argument is set to 0 or DTMF digit detection to insensitivity, the neglect probability is decreasing and the detection error probability is increasing as the DTMF digit collection tolerance becomes larger. DTMF digit detection is implemented by calculating the spectrum of the input voice signal. Its spectrum shape is restricted to the configured thresholds. A DTMF dial tone is regarded valid only when all the constraints are met. To understand this, you must be aware of DTMF dial tone, as shown in the following figure: 1-62 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Figure 1-1 Spectrum of keys The tone of each telephone digit is composed of two single-frequency tones. For example, the tone of the digit 1 is compounded by two sine wave signal tones of 697 Hz and 1209 Hz. A valid key tone must last at least 45 milliseconds and have an inter-digit delay of 23 milliseconds. Refer to ITU-T Q.24 recommendation for full information. Figure 1-2 Spectrum of key 1 Figure 1-2 illustrates the spectrum of key 1. Compared with other frequency points, the energy levels at frequency points of 697 Hz and 1209 Hz are relatively greater. The underlying layer DSP regards the frequencies of 1209 Hz, 1336 Hz, 1477 Hz, and 1633 Hz as column frequency group and the frequencies of 697 Hz, 770 Hz, 852 Hz, and 941Hz as row frequency group. Each DTMF key tone is composed of one column frequency and one row frequency. The DSP module determines whether the input voice signal is a valid DTMF digit by its energy at the above eight frequencies and their double frequencies. The maximum energy in the row frequency group is ROWMAX, its double energy is ROW2nd; the maximum energy in the column frequency group is COLMAX and its double energy is COL2nd. 1-63 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Example # Set DTMF threshold 9 in voice subscriber-line view. [H3C-voice-line0/0/0] dtmf threshold analogue 9 30 # Restore the default of DTMF threshold 9 in voice subscriber-line view. [H3C-voice-line0/0/0] undo dtmf threshold analog 9 1.1.51 echo-canceller Syntax echo-canceller { enable | tail-length milliseconds | parameter { convergence-rate value | max-amplitude value | mix-proportion-ratio value | talk-threshold value } } undo echo-canceller { enable | tail-length | parameter } View Voice subscriber-line view Parameter enable: Enables echo-cancellation function. By default, the function is enabled. tail-length milliseconds: Adjusts echo-cancellation duration, that is, the time that elapse when a subscriber hears the voice echoed back. It ranges from 0 to 64 ms and defaults to 32 ms. parameter: sets echo-cancellation parameters. convergence-rate value: Sets the ascending rate of comfort noise amplitude. It ranges from 0 to 511 and defaults to 255. The greater the value, the quicker the convergence. If a subscriber hears echoes after speaking, or the background noise of the peer is loud, you can increase this value. Note that if this value is too great, the noise may be not smooth enough. max-amplitude value: Sets maximum amplitude of comfort noise. It ranges from 0 to 10 and defaults to 0. The higher the value, the greater the maximum noise amplitude. A value of 0 indicates that the system performs only nonlinear process and does not add comfort noise. When the environmental noise is louder, you can increase this value. Note that if this value is too great, the noise may be not smooth enough. mix-proportion-ratio value: Sets comfort noise mixture proportion control factor. It ranges from 1 to 3,000 and defaults to 100. The greater the value, the higher the proportion of noise in the hybrid of noise and voice. If echoes occur when only one subscriber is speaking, you can increase this value. Note that if this value is too great, the voice may be desultory. talk-threshold value: Sets the threshold of simultaneous talks. It ranges from 1 to 2 and defaults to 1. If echoes occur when the two parties speak simultaneously, you may 1-64 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands increase this value. Note that if this value is too great, the filter factor convergence rate may become slow. Description Use the echo-canceller command to configure echo-cancellation parameters. Use the undo echo-canceller command to remove the configuration. During a call, the voice of a subscriber may be echoed back to the handset. This is because analog voice signals are leaked into the reception path of the subscriber. You can use the echo-cancellation function provided by the voice gateway to solve this problem to a certain extent. If a too large time value is set, the converge time of the echo canceling on the network will become longer, so when the connection has just been established, the user may hear the echo; and if the time value is set as too small, the user may also hear part of the echo, because the relatively longer echo has not been completely cancelled. There are no echo and echo-cancel on the IP side. Signal leakage occurs only on the analog circuit part of the voice call path. Digital networks do not suffer from this problem. Note: The echo-canceller enable command must be used in pair with the echo-canceller tail-length command. Related command: subscriber-line. Example # Set the echo-cancel sampling time as 24 ms on subscriber line 1/0/0. [H3C-voice-line1/0/0] echo-canceller enable [H3C-voice-line1/0/0] echo-canceller tail-length 24 1.1.52 em-phy-parm Syntax em-phy-parm { 2-wire | 4-wire } undo em-phy-parm { 2-wire | 4-wire } View Voice subscriber-line view 1-65 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter 2-wire: Choose the 2-wire analog E&M wire scheme. 4-wire: Choose the 4-wire analog E&M wire scheme. Description Use the em-phy-parm command to configure a wire scheme for the analog E&M subscriber-line. Use the undo em-phy-parm command to cancel an existing wire selection scheme. By default, the 4-wire analog E&M wire scheme is selected. This command is only applicable to analog E&M subscriber-line. When analog E&M is used in PBX communication, its voice uses two or four wires, plus two or four signaling wires. So, a 4-wire analog E&M actually has at least six wires. 2-wire mode provides full-duplex voice transmission, in which voice is transmitted between two wires in both directions. 4-wire mode resembles the simplex mode. Every two wires are responsible for voice transmission in one direction. This command is used for the wire match between the voice router and remote end equipment connected with it. It has influence on voice transmission only and has nothing to do with signaling. Example # Choose the 4-wire scheme for the analog E&M subscriber-line. [H3C-voice-line1/0/0] em-phy-parm 4-wire 1.1.53 em-signal Syntax em-signal { delay | immediate | wink | passthrough } undo em-signal { delay | immediate | wink } View Voice subscriber-line view Parameter wink: Wink start mode. The caller end hooks off to seize the line through line E, and it has to wait for a wink signal from the remote end before sending out the called number. immediate: Immediate start mode. The caller end hooks off to seize the line through line E and sends the called number. The prerequisite for using the immediate start mode is: The equipment at the remote end should listen to the dial signal immediately after identifying the off-hook signal. 1-66 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands delay: When using the delay start mode, the calling end occupies the trunk line, and the called end, such as PBX, will also enter the hook-off state to respond the caller till it is ready for receiving the called number. passthrough: Converts the physical signals received from the E line into IP packets, transfers them transparently from the local gateway to the peer gateway via an VoIP network (currently, only H.323 VoIP network), reduces them to physical signals, and outputs them from the M line. Description Use the em-signal command to configure a voice subscriber-line start mode. Use the undo em-signal command to cancel the set voice subscriber-line start mode. By default, analog E&M subscriber-lines select the immediate start mode. This command is only applicable to analog E&M subscriber-line. The start mode used for an analog E&M subscriber-line should be in consistency with the PBX connected with it. When using the immediate start mode, the numbers will not be correctly sent or received due to the signaling type in some PBX, please modify start mode as the wink or delay. The em-signal passthrough command will make the device transparently transfer E&M signals without any processing. In this case, the start mode is immediate start. The em-signal passthrough command together with the open-trunk command can implement transparent transmission of E&M signals as well as a permanent E&M connection. Related command: delay. Example # Configure the immediate mode for the analog E&M subscriber-line. [H3C-voice-line1/0/0] em-signal immediate 1.1.54 entity Syntax entity entity-number { voip | pots | vofr } undo entity { entity-number | all | voip | pots | vofr } View Voice dial program view Parameter entity-number: Identify a voice entity, The value range is from 1 to 2147483647. 1-67 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands all: All dial program entities. voip: Key character, indicating that this subscriber-line is a network subscriber-line. pots: Key character, indicating that this subscriber-line is an often used telephone subscriber-line. Description Use the entity command to configure a voice entity and enter its view (at the same time specify the working mode related to voice). Use the undo entity command to cancel an existing voice entity. In a global view, use the entity command to enter a Voice entity view, and use quit to return to the dial program view. Related command: line. Note: The entity-number assigned to a VoIP or POTS entity must be unique among all VoIP and POTS entities. Example # Create and enter the Voice entity view to configure a POTS voice entity whose identification is 10. [H3C-voice] dial-program [H3C-voice-dial] entity 10 pots 1.1.55 fast-connect Syntax fast-connect undo fast-connect View VoIP voice entity view Parameter None Description Use the fast-connect command to enable fast connect. 1-68 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Use the undo fast-connect command to disable fast connect. By default, fast connect is disabled. As there is no ability negotiation for fast connect, the ability confirmation of the two parties is determined by the called gateway. When the router acts as a calling gateway, one can set whether or not to apply fast connect mode for each originated call. If the calling gateway adopts fast connect mode, the called gateway will adopt it, too. Otherwise, neither one will do so. Fast connect procedure will be used when both the calling and called parties support fast connect. Provided that neither the calling nor the called gateway supports fast connect mode, the system will automatically switch to normal connect procedure to resume the call. It is OK to only configure fast-connect command for VoIP voice entity on the calling gateway. Just after successfully enabling the fast connect can the tunnel function be configured. Related command: outband, tunnel-on, voip call-start. Example # Enable fast connect for VoIP voice entity 10. [H3C-voice-dial-entity10] fast-connect 1.1.56 first-rule Syntax first-rule rule-number undo first-rule View Voice number-substitute view Parameter rule-number: Serial numbers of the number substitution rules, ranging from 0 to 127. Description Use the first-rule command to configure the number substitution rule that is first used in the current number substitution list. Use the undo first-rule command to restore the default number substitution rule. By default, the configured rule with the smallest serial number is used first. In a voice call, the system first uses the rule that is defined by the first-rule command when it begins to use the number substitution rules. If this rule fails, it will try all other rules in order, till it finds the one that works or till it confirms all the rules do not work. 1-69 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Example # Set rule 4 as the first-used rule in number substitution list 20. [H3C-voice-dial] number-substitute 20 [H3C-voice-dial-substitute20] rule 4 663 3 [H3C-voice-dial-substitute20] first-rule 4 1.1.57 first-rule Syntax first-rule rule-number undo first-rule View Voice number-substitute view Parameter rule-number: Serial numbers of the number-substitute rules, ranging from 0 to 127. Description Use the first-rule command to configure the number-substitute rule that is first used in the current number-substitute list. Use the undo first-rule command to restore the default number-substitute rule. By default, the configured rule with the smallest serial number is used first. In a voice call, the system first uses the rule that is defined by the first-rule command when it begins to use the number-substitute rules. If this rule fails, it will try all other rules in order, till it finds the one that works or till it confirms all the rules do not work. Example # Set rule 4 as the first-used rule in number-substitute list 20. [H3C-voice-dial] number-substitute 20 [H3C-voice-dial-substitute20] rule 4 663 3 [H3C-voice-dial-substitute20] first-rule 4 1.1.58 hookoff-time Syntax hookoff-time time undo hookoff-time 1-70 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View FXO voice subscriber-line view Parameter time: Length of the hangup timer, in the range 60 to 36000 seconds. By default, no hangup timer is set. Description Use the hookoff-time command to configure the hangup timer. When this timer times out, the interface hangs up. Use the undo hookoff-time command to cancel the setting of the hangup timer. In some countries, PBXs do not play busy tone; if they play, the busy tone only lasts a short period of time. When noise is present on a transmission link, the silence-th-span command may be inadequate for solving the problem that the FXO interface does not hang up. In this case, you may use the hookoff-time command to address the problem. Once configured the hookoff-time command takes effect on all interfaces on the card. Example # Set the hangup timer to 500 seconds. [H3C-voice-line3/0/0] hookoff-time 500 1.1.59 impedance Syntax impedance { country-name | R550 | R600 | R650 | R700 | R750 | R800 | R850 | R900 | R950 } undo impedance View FXO voice subscriber-line view Parameter country-name: Specifies a country so that its impedance standard is used. It can be Australia, Austria, Belgium-Long, Belgium-Short, Brazil, China, Czech-Republic, Denmark, ETSI-Harmanized, Finland, France, German-Swiss, Greece, Hungary, India, Italy, Japan, Korea, Mexico, Netherlands, Norway, Portugal, Slovakia, Spain, Sweden, U.K., US-Loaded-Line, US-Non-Loaded, or US-Special-Service. R550: 550-ohm real impedance. R600: 600-ohm real impedance. 1-71 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands R650: 650-ohm real impedance. R700: 700-ohm real impedance. R750: 750-ohm real impedance. R800: 800-ohm real impedance. R850: 850-ohm real impedance. R900: 900-ohm real impedance. R950: 950-ohm real impedance. Description Use the impedance command to configure the current electric impedance on a voice subscriber-line. Use the undo impedance command to restore the default electric impedance on the voice subscriber-line. You can specify an impedance value by specifying the country where the value applies. You may just input the leading letters that uniquely identify the country however. The default electric impedance on the voice line is China. Example # Configure the current electric impedance to r600 on voice subscriber-line 1/0/0. [H3C-voice-line1/0/0] impedance r600 1.1.60 line Syntax line line-number undo line View POTS voice entity view Parameter line-number: Number of a subscriber line. Description Use the line command to associate the voice entity with a specified voice subscriber-line. Use the undo line command to cancel this association. This command can be used in POTS Voice entity view only. It can take effect only on FXS, FXO, analog E&M, and digital E&M interfaces. 1-72 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands solt 3 solt 2 solt 0 slot 1 line0/0/0 line0/0/1 line0/0/2 line0/0/3 line1/0/0 line1/0/1 Figure 1-3 Number the voice subscriber-lines on voice cards The above figure displays the rear view of a router installed with four voice cards. The first card provides two subscriber lines, while other three provides four subscriber lines each. The voice subscriber-line numbers are set according to the card sequence from the left to the right, and add 1 in order starting from 0. Example # Associate voice entity 10 and voice subscriber-line 0/0/0. [H3C-voice-dial-entity10] line 0/0/0 1.1.61 match-template Syntax match-template match-string undo match-template View Voice entity view Parameter match-string: Match template. Its format is [ + ] { string [ T ] [ $ ] | T }, with the largest length of 31 characters. The characters are described in the following. z +: Appears at the beginning of a calling number to indicate that the number is E.164-compliant. z $: Is the last character, indicating the end of the number. That means the entire called number must match all the characters before “$” in the string. z T: Timer. It means the system is waiting the subscriber for dialing any number till: the number length threshold is exceeded; the subscriber inputs the terminator; or the timer expires. It seems to subscribers that T matches any number in any length. 1-73 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands string: A string composed of any characters of “0123456789ABCD#*.!+%[]()-”. z The meanings of the characters are described in the following table: Table 1-22 Meanings of the characters in string Character Meaning 0-9 Numbers from 0 to 9. Each means a digit. ABCD Each character means a digit. # and * Each means a valid digit. . A wildcard. It can match any digit of a valid number. For example, 555. . . . matches any string that begins with 555 and with four additional characters. ! The character or characters right in front of it does not appear or appears once. For example, 56!1234 can match 51234 and 561234. + The character or characters right in front of it appears once or several times. But its appearance at the beginning of the whole number means the number is E.164-compliant. - Hyphen. It connects two values (the smaller one before it and the bigger one after it) to indicate a range. For example, “1-9” means numbers from 1 to 9 (inclusive). % The character or characters right in front of it does not appear, or appears several times. [] Select one character from the group. For example, [1-36A] can match only one character among 1, 2, 3, 6, and A. () A group of characters. For example, (123A) means a string “123A”. 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 at least appear once. Note: z The character or characters in front of "!”, “%”, and “+” are not to be matched accurately. They are handled similar to the wildcard “.”. Moreover, these symbols cannot be used alone. There must be a valid digit or digits in front of them. z If you want to use “[ ]” and “( )” at the same time, you must use them in the format “( [ ] )”. Other formats, such as “[ [ ] ]” and “[ ( ) ]” are illegal. z “-“ can only be used in “[ ]”, and it only connects the same type of characters, such as “0-9”. The formats like “0-A” are illegal. Description Use the match-template command to configure the match template for a voice entity. 1-74 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Use the undo match-template command to delete this configuration. The match template defined by the match-template command can be used to match the number reaching the corresponding voice entity. The voice entity will complete the call if the match is successful. The match template can be defined flexibly. It can not only be a string of a unique number like 01016781234, but also an expression that can match a group of numbers, such as “010[1-5]678…”. They are used to match the actual numbers in the received call packets to complete the calls. Note: z Comware software does not check the validity of E.164 numbers. z In E1 voice, “T”, “#”, and “*” are not supported currently. Example # Set 5557922 as the telephone number of voice entity 10. [H3C-voice-dial-entity10] match-template +5557922 # Set 66.... as the match template of voice entity 20. [H3C-voice-dial-entity20] match-template 66.... # Set the match template for the numbers beginning with 661, 662, 663, and 669, and containing four other digits. [H3C-voice-dial-entity1] match-template 66[1-39].... # Set the match template for the numbers beginning with 66 and 6602 and containing four other digits. [H3C-voice-dial-entity1] match-template 66(02)!.... # Set the match template for the numbers beginning with 66 and within the length of 31 digits. [H3C-voice-dial-entity1] match-template 66T # Set the match template for the numbers beginning with any number and within the length of 31 digits. [H3C-voice-dial-entity1] match-template T 1.1.62 max-call (voice dial program view) Syntax max-call set-number max-number undo max-call { all | set-number } 1-75 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View Voice dial program view Parameter set-number: Specify a max-call set. Its value ranges from 1 to 2147483647. You can configure 128 sets at most. max-number: Specify the maximum number of call connections for a max-call set. It ranges from 1 to 120. All: All max-call sets. Description Use the max-call command to configure max-call sets (128 sets at most). Use the undo max-call command to delete the specified set or all the max-call sets. This command is used to limit the number of call connections for a voice entity or a group of voice entities. It needs to be used with the max-call command in voice entity view: It defines the serial number of a max-call set and the maximum call connections; while the max-call command in voice entity view binds that voice entity to the max-call set with the serial number. Related command: max-call (voice entity). Example # Set the maximum number of call connections of max-call set 1 to 5. [H3C-voice-dial] max-call 1 5 1.1.63 max-call (voice entity view) Syntax max-call set-number undo max-call View Voice entity view Parameter set-number: Specifies a max-call set. Its value ranges from 1 to 2147483647. Description Use the max-call command to bind a voice entity to the max-call set specified by set-number. 1-76 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Use the undo max-call command to delete the binding. Each voice entity can only be bound to one max-call set, but you can change the binding. By default, no max-call set is bound. That means there is no limitation on the number of call connections. This command is used with the max-call command in voice dial program view: the max-call command in voice dial program view sets the serial number of max-call set and the maximum number of call connections; this command binds the voice entity to the max-call set with the serial number. Related command: max-call (voice dial program). Example # Bind voice entity 10 to max-call set 1. [H3C-voice-dial-entity10] max-call 1 1.1.64 normal-connect slow-h245 Syntax normal-connect slow-h245 undo normal-connect slow-h245 View VoIP voice entity view Parameter None Description Use the normal-connect slow-h245 command to configure disabling the calling end from actively initiating an H245 connection request to the called end before the called end is hooked off in voice entity view. Use the undo default entity normal-connect slow-h245 command to restore the default value (that is, to allow the calling end to actively initiate an H245 connection request to the called end before the called end is hooked off). By default, the calling end is allowed to actively initiate an H245 connection request to the called end before the called end is hooked off. Use the normal-connect slow-h245 command to disable the calling end from actively initiating an H245 connection request to the called end before the called end is hooked off in voice entity view. Related command: fast-connect, default entity normal-connect slow-h245. 1-77 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Example # Configure for VoIP voice entity 103 to disable the calling end from actively initiating the H245 connection request to the called end before the called end is hooked off. [H3C-voice-dial-entity103]normal-connect slow-h245 1.1.65 number-match Syntax number-match { longest | shortest } undo number-match View Voice dial program view Parameter longest: Indicates to perform the match according to the longest number. shortest: Indicates to perform the match according to the shortest number. Description Use the number-match command to configure a global number match-policy. Use the undo number-match command to restore the default. By default, match according to the shortest number. Command number-match is used to decide if the match is performed according to the longest or the shortest number for the number match. For instance, match-template 0106688 and match-template 01066880011 are respectively configured in two voice entities. when the user dials 01066880011, if the router is configured with the shortest number match-policy, then the router originates a connection to 0106688 at the remote end, and the four numbers 0011 will not be processed; if the router is configured with the longest number match-policy, and the user only dials 0106688, the router will wait for the user to dial. After timeout, the number match-policy configured with the system is neglected, and the shortest number match-policy will be automatically followed to make a call; if the user dials 0106688# (here the "#” represents the dial terminator configured by the system), the router will likewise neglect the number match-policy configured by the system and use the shortest number match-policy, thus providing a greater flexibility for the configuration of user dial scheme. Related command: match-template. Example # Configure that the number match is performed according to the longest number. [H3C-voice-dial] number-match longest 1-78 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.66 number-substitute Syntax number-substitute list-number undo number-substitute { list-number | all } View Voice dial program view Parameter list-number: Serial number of number substitution rule list. Its value ranges from 1 to 2147483647. all: All number substitution rule lists. Description Use the number-substitute command to create a number substitution rule list and enter voice dial program view. Use the undo number-substitute command to delete the specified number substitution rule list or all the number substitution rule lists. By default, no number substitution rule list is created. Related command: rule and substitute. Example # Enter voice dial program view and create a number substitution rule list. [H3C-voice-dial] number-substitute 1 [H3C-voice-dial-substitute1] 1.1.67 open-trunk Syntax open-trunk { caller [ monitor interval ] | called } undo open-trunk [caller monitor ] View Analog E&M voice subscriber line view Parameter caller: Enables E&M non-signaling mode but not the monitoring function when voice gateway works as the calling end. 1-79 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands monitor interval: Enables E&M non-signaling mode and the monitoring function at the same time when voice gateway works as the calling end. The monitoring time is specified by interval. called: Enables E&M non-signaling mode when voice gateway works as the called end. Description Use the open-trunk command to enable E&M non-signaling mode. Use the undo open-trunk command to delete E&M non-signaling mode. By default, E&M non-signaling mode is disabled.  Note: When you use the open-trunk caller monitor interval command to enable E&M non-signaling mode and the monitoring function at the same time, execute the undo open-trunk monitor command to delete the monitoring function (Note that the execution of this command only disables the monitoring function but does not delete E&M non-signaling mode.); If you use the undo open-trunk command, not only E&M non-signaling mode will be deleted, but also the monitoring function will be disabled. Example # Configure to enable E&M non-signaling mode and the monitoring function at the same time. The monitoring time is 120 seconds. [H3C-voice-line3/0/0] open-trunk caller monitor 120 1.1.68 outband Syntax outband { h225 | h245 | sip | vofr } undo outband View Voice entity view Parameter h225: Enables DTMF H.225 out-of-band transmission. h245: Enables DTMF H.245 out-of-band transmission. sip: Enables DTMF SIP out-of-band transmission. 1-80 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands vofr: Enables DTMF VoFR out-of-band transmission. Description Use the outband command to configure transmission of DTMF code in the outband mode. Use the undo outband command to restore transmission of DTMF code to inband mode. By default, the inband transmission mode is adopted. Once transmitting DTMF code in outband mode has been configured for NON-STANDARD router, in order to confirm whether or not the peer gateway can accept such transmission mode of DTMF, the ability negotiation must be fulfilled after the connection is established. Currently, H3C series routers only reactively accept transparent transmission ability negotiation of DTMF code. In no case will it actively initiate the negotiation process mentioned above. In order to transparently transmit DTMF code, it is demanded, during actual application, to configure this command for VoIP voice entity on the calling gateway and also for POTS voice entity for the called gateway. Related command: fast-connect and tunnel-on. Note: z In VoIP entity view, only SIP, h225, and h245 DTMF out-of-band transmission modes can be displayed. z In VoIP entity view, if the routing is set to IP or RAS, only h.225 and h.245 DTMF out-of-band transmission modes can be displayed, while if the routing mode is set to SIP, only the sip DTMF out-of-band transmission mode can be displayed. z In POTS entity view, all of the five DTMF transmission modes h225, h245, sip and vofr can be displayed. z In VoIP entity view, if the non-fast connection mode is adopted, DTMF can be directly transmitted through H.245 or H.225 protocol in outband mode, while if fast connection mode adopted, DTMF can only be transmitted in outband mode through H.225 protocol. Example # Configure DTMF code outband transmission in the fast connect mode for VoIP voice entity 10. [H3C-voice-dial-entity10] fast-connect [H3C-voice-dial-entity10] outband h245 1-81 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands # Configure the NTE DTMF out-of-band transmission for VoIP entity 10. [H3C-voice-dial-entity10] outband nte 1.1.69 overlap timer Syntax overlap timer seconds undo overlap timer View Voice dial program view Parameter seconds: Wait time in seconds for a call origination, in the range of 0 to 10 with a default of 10. The value “0” means that a call is originated immediately. Description Use the overlap timer command to configure the wait time for a call origination in the case that a match template with a T sign is configured on the terminating gateway in the H.323 overlap mode. Use the undo overlap timer command to restore the default wait time. Example # Set the wait time for a call origination to 5 seconds in the case that a match-template with a T sign is configured on the terminating gateway in the H.323 overlap mode. [H3C-voice-dial-entity2000] overlap timer 5 1.1.70 overlap voip h323 Syntax overlap voip h323 undo overlap voip h323 View Voice dial program view Parameter None 1-82 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the overlap voip h323 command to configure the calling party to support h323 overlap mode for sending digits. Use the undo overlap voip h323 command to restore the default configuration. By default, h323 overlap mode is not supported by the calling party. Example # Configure the calling party to support h323 overlap mode for sending digits. [H3C-voice-dial]overlap voip h323 1.1.71 payload-size Syntax payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } time-length undo payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } View Voice entity view Parameter g711: Specifies the time length of voice packets with g711 coding (g711alaw or g711ulaw). It ranges from 10 to 30 milliseconds and defaults to 20 milliseconds. g723: Specifies the time length of voice packets with g723 coding (g723r53 or g723r63). It ranges from 30 to 180 milliseconds and defaults to 30 milliseconds. g726r16: Specifies the time length of voice packets with g726r16 coding. It ranges from 10 to 110 milliseconds and defaults to 30 milliseconds. g726r24: Specifies the time length of voice packets with g726r24 coding. It ranges from 10 to 70 milliseconds and defaults to 30 milliseconds. g726r32: Specifies the time length of voice packets with g726r32 coding. It ranges from 10 to 50 milliseconds and defaults to 30 milliseconds. g726r40: Specifies the time length of voice packets with g726r40 coding. It ranges from 10 to 40 milliseconds and defaults to 30 milliseconds. g729: Specifies the time length of voice packets with g729 coding (g729r8 or g729a). It ranges from 20 to 180 milliseconds and defaults to 30 milliseconds. time-length: Duration for DSP to assemble a packet in the corresponding codec mode. 1-83 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the payload-size command to notify the underlying layer how much time DSP spends assembling a voice packet, or the time length of each voice packet. Use the undo payload-size command to restore the default. This time length may differ by card type. Because voice entities, which are upper layer relevant, are hardware independent, the upper-layer can only verify whether the specified value is in the valid range for the applied codec mode. Whether this value is valid within the range is decided by DSP based on hardware type. If valid, the value delivered by the upper layer is used; if otherwise, the default value for the codec mode is used. If you find that the configured duration does not take effect, first check that the assigned value is valid for the card type and codec mode. Table 1-23 Valid ranges corresponding to cards Codec mode Valid range for C55xx (milliseconds) G711 G723 G729 10 to 30 30 to 180 20 to 180 (must be an integral multiple of 10) (must be an integral multiple of 30) (must be an integral multiple of 10) 30 to 180 20 to 180 (must be an integral multiple of 30) (must be an integral multiple of 10) 30 20 Valid range for C54xx (milliseconds) Default (milliseconds) 20 20 Description 8FXS VG2032 VG2016 4VI/2VI/17VI VG1040 VG1041 –– In voice dial program view, you can use the default entity command to configure global attributes for voice entities, namely the default time that DSP spends assembling a packet in each codec mode. Related command: default entity, entity compression. Example # Set the time that DSP spends assembling a voice packet in g711 mode to 30 milliseconds for voice entity 8801. [route-voice-dial-entity8801] payload-size g711 30 # Restore the default time that DSP spends assembling a voice packet in g711 codec mode for voice entity 8801. [route-voice-dial-entity8801] undo payload-size g711 1-84 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands # Set the default time that DSP spends assembling a voice packet in g729 mode to 180 milliseconds. [route-voice-dial] default entity payload-size g729 180 # Restore the initial default time that DSP spends assembling a packet in g729 codec mode in voice dial program view. [route-voice-dial] undo default entity payload-size g729 1.1.72 plc-mode Syntax plc-mode { general | specific } undo plc-mode View Voice subscriber line view Parameter general: Uses the universal frame erasure algorithm. specific: Uses the specific algorithm provided by the voice gateway. Description Use the plc-mode command to configure packet loss compensation mode. Use the undo plc-mode command to restore the default packet loss compensation mode. By default, the specific algorithm is used. This command takes effect only on FXO and FXS interfaces. Example # Configure the voice gateway to use the universal packet loss compensation algorithm. [H3C-voice-line1/0/0] plc-mode general 1.1.73 priority Syntax priority priority-order undo priority View Voice entity view 1-85 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter priority-order: Priority of a voice entity. Its value ranges from 0 to 10. The smaller the number is, the higher the priority is. That means 0 is the highest priority and 10 is the lowest priority. Description Use the priority command to configure the priority levels for voice entities. Use the undo priority command to restore the default priority level. By default, the priority level is set to 0. If you have configured priority levels for voice entities and have configured priority in voice entity select rule (see select-rule), the system will first select the voice entity of the highest priority when it initiates a call. If the voice entity of the highest priority fails, it will try those of lower priority levels to initiate the call. Example # Set the priority level of voice entity 10 to 5. [H3C-voice-dial-entity10] priority 5 1.1.74 private-line Syntax private-line string [ if-match no-called-num ] undo private-line View Voice subscriber-line view Parameter string: E.164 telephone number of the destination end, and it may include these characters: 0 to 9, *, and #. if-match: Specifies the condition under which the private auto-ring mode takes effect. no-called-num: Vacant number. Description Use the private-line command to configure private auto-ring mode for the subscriber line and the E.164 telephone number of the destination end. Use the undo private-line command to cancel the specified connection mode. By default, no private auto-ring mode is configured. 1-86 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands This command is applicable to FXO, FXS, analog E&M subscriber-lines and digital E1 voice subscriber-line. The private-line command is used to specify a connection mode for subscriber-line. The parameter string will serve as the called number of all the calls incoming to this subscriber line, i.e., after off-hook, the user need not perform any operation and the system will dial out the string as the called number automatically. The command option if-match no-called-num indicates that the private auto-ring mode will be started only when the called number is vacant. If the private-line command is not configured, when the subscriber-line enters an off-hook status, the standard session application program will generate a dial tone until enough numbers are collected and the call process is completed. Note: The command option if-match no-called-num only applies to the voice subscriber lines using PRI signaling. Example # Set an automatic dialing to 5559262 after off-hook on the subscriber line 1/0/0. [H3C-voice-line1/0/0] private-line 5559262 1.1.75 progress-tone Syntax progress-tone { local | remote | none } undo progress-tone View Voice subscriber line view Parameter local: Specifies the local end (namely, the DSP) to play the progress tone. remote: Specifies the remote end of the trunk (namely, the PBX) to play the progress tone. none: Specifies the remote end of the trunk or the originating end of an IP call, instead of local end (DSP) to play the progress tone. 1-87 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the progress-tone command to specify the player of the progress tone for an IP call from the voice trunk perspective. Use the undo progress-tone command to restore the default player of the progress tone. After the progress-tone none command is configured, if the local end receives the progress tone from the remote end of the trunk, the remote end will plays the tone. Otherwise the local end plays the tone. By default, the local end plays the progress tone. Example # Configure the voice subscriber line 1/0:30 to use the progress tone played by the remote end of the trunk. [H3C-voice] subscriber-line 1/0:30 R2 interface encountered [H3C-voice-line1/0:30] progress-tone remote 1.1.76 receive gain Syntax receive gain value undo receive gain View Voice subscriber-line view Parameter value: Voice input gain ranging from -14.0 to 14.0 in dB with one digit after the decimal point. By default, the value is 0 dB. Description Use the receive gain command to configure the gain value at the voice subscriber-line input end. Use the undo receive gain command to restore the default value. This command is applicable to FXO, FXS, analog E&M subscriber-lines and digital E1 voice subscriber-line. When the voice signal on the line attenuates to a relatively great extent, this command can be used to appropriately enhance the voice input gain. 1-88 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Caution: Adjusting the gain may cause voice calls unable to be established. Therefore, when required, adjust the gain only under the direction of technical support staff. Related command: transmit gain and subscriber-line. Example # Configure the voice input gain as 3.5dB on subscriber line 1/0/0. [H3C-voice-line1/0/0] receive gain 3.5 1.1.77 reset voice call-history-record line Syntax reset voice call-history-record line View User view Parameter None Description Use the reset voice call-history-record line command to clear the call history of all voice subscriber lines, or information displayed by call-history-record line command. Example # Clear the call history of all voice-subscriber lines. reset voice call-history-record line Really reset all of the call-history information? [y]y All of the call-history information have been removed! 1.1.78 reset voice ipp Syntax reset voice ipp View User view 1-89 the display voice Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter None Description Use the reset voice ipp command to reset IPP statistics. Related command: display voice vpp. Example # Clear IPP statistics. reset voice ipp 1.1.79 reset voice rcv Syntax reset voice rcv View User view Parameter None Description Use the reset voice rcv command to reset RCV statistics. Related command: display voice rcv statistic. Example # Clear RCV statistics. reset voice rcv 1.1.80 reset voice vpp Syntax reset voice vpp View User view Parameter None 1-90 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the reset voice vpp command to reset VPP statistics. Related command: display voice vpp. Example # Clear VPP statistics. reset voice vpp channel 1 1.1.81 reset voice voip data-statistic Syntax reset voice voip data-statistic View User view Parameter None Description Use the reset voice voip data-statistic command to reset statistics information of voice data. After voice data statistics is enabled via the vqa data-statistic command, all statistics items (time for successfully searching the voice table, total number of received data packets, time for searching the table in fast and common modes and voice and fax of receive and transmit channels) perform data accumulation until all counters are cleared via the reset voice voip data-statistic command. After that, new statistics begins. Related command: vqa data-statistic and display voice voip data-statistic. Example # Clear statistics information of voice data. reset voice voip data-statistic 1.1.82 rule 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 } 1-91 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View Voice number-substitute view Parameter all: Deletes all number substitution rules. rule-tag: Number of a rule. Its value ranges from 0 to 127. input-number: Input string for number substitute. Format of the number is [ ^ ] [ + ] string [ $ ], with the maximum length of 31 digits. The characters are described in the following: ^: Indicates a number must be matched from the first character. When the system z matches a subscriber number with the match string, it must begin from the first character of the subscriber number. +: Appears at the beginning of a calling number to indicate that the number is z E.164 compliant. $: Indicates that the last character of the number must be matched, that is, the last z character of the number must match the last character of the match string. string: String composed of any characters of “0123456789#*.!%”. The characters z are described in the following table: Table 1-24 Meanings of the characters in string Character Meaning 0-9 Numbers from 0 to 9. Each means a digit. # and * Each means a valid digit. . A wildcard. It can match any digit of a valid number. For example, 555. . . . matches any string that begins with 555 and with four additional characters. ! The character or characters right in front of it does not appear or appears once. For example, 56!1234 can match 51234 and 561234. + The character or characters right in front of it appears once or several times. % The character or characters right in front of it does not appear or appears several times. output-number: Output string for number substitute. It is composed of any characters of “0123456789#*.”, with the largest length of 31 digits. The meanings of the characters are shown in the table above. number-type: Number type. numbering-plan: Numbering plan. 1-92 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands input-number-type: Input type number, the value of which are abbreviated, any, international, national, network, reserved, subscriber, and unknown. z abbreviated: Abbreviated number type. z any: It is not a specific number type, meaning any input number type can match. z international: International number type. z national: National number type. z network: Network-specific number type. z reserved: Reserved for extension. z subscriber: Subscriber number type. z unknown: Unknown number type. output-number-type: Output number type, refer to the argument input-number-type for its value, excluding any. z abbreviated: Abbreviated number type. z international: International number type. z national: National number type. z network: Network-specific number type. z reserved: Reserved for extension. z subscriber: Subscriber number type. z unknown: Unknown number type. input-numbering-plan: Input numbering plan, the value of which are any, isdn, national, private, reserved, telex, and unknown. z any: It is not a specific numbering plan, meaning any input numbering plan can match. z data: Data numbering plan. z isdn: ISDN telephone numbering plan. z national: National standard numbering plan. z private: Private numbering plan. z reserved: Reserved for extension. z telex: Subscriber telex numbering plan. z unknown: Unknown numbering plan. output-numbering-plan: Output numbering input-numbering-plan for its value, excluding any. z data: Data numbering plan. z isdn: ISDN telephone numbering plan. z national: National standard numbering plan. z private: Private numbering plan. z reserved: Reserved for extension. z telex: Subscriber telex numbering plan. z unknown: Unknown numbering plan. Pay attention to the following points: 1-93 plan, refer to the argument Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands The character or characters in front of "!”, “%”, and “+” are not to be matched accurately. They are handled similar to the wildcard “.”. Moreover, these symbols cannot be used alone. There must be a valid digit or digits in front of them. Dots in input-number and output-number are handled in three ways: z Treat dots in output-number as invalid. When you use the dot-match command to configure the dot match rule as end-only (that is, only dots at the end of the input format are handled), the dots in output-number are discarded immediately, and the digits corresponding to all the dots at the end of input-number are added to the end of output-number. z Discards extra dots in output-number. When you use the dot-match command to configure the dot match rule as right-left (from right to left) or left-right (from left to right), and the dot digits in output-number are more than those in input-number, all digits corresponding to the dots in input-number are selected to replace the dots in output-number one by one from left to right. The dots not replaced in output-number are discarded. That is, the dots to the right are discarded. z Discards (the digits corresponding to) the extra dots in input-number. When you use the dot-match command to configure the dot match rule as right-left (from right to left) or left-right (from left to right), and the number of dot digits in input-number are greater than or equal to that in output-number, two cases exist: Case 1: For right-left matching, extract from right to left the digits corresponding to the dots in input-number and use them to replace the dots in output-number one by one. The digits corresponding to dots that are not extracted in input-number are discarded. Case 2: For left-right matching, extract from left to right the digits corresponding to the dots in input-number and use them to replace the dots in output-number one by one. The digits corresponding to dots that are not extracted in input-number are discarded. Description Use the rule command to configure the number substitution rule. Use the undo rule command to delete the specified or all number substitution rules. By default, no number substitution rule is configured. When you have successfully created a number substitution rule list, you need to use the command to configure the number substitution rules in it. Related command: substitute, number-substitute, first-rule, and dot-match. Example # Configure the number substitution rules for number substitution list 1 [H3C-voice-dial-substitute1] rule 1 ^91 1 [H3C-voice-dial-substitute1] rule 2 ^92 2 number-type any international numbering-plan any private 1-94 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands [H3C-voice-dial-substitute1] rule 3 ^93 3 number-type any international numbering-plan any isdn 1.1.83 select-rule rule-order Syntax select-rule rule-order 1 [ 2 [ 3 ] [ 4 ]] undo select-rule rule-order View Voice dial program view Parameter The following lists the meanings of the serial numbers. 1: Defines the voice entity select rule as exact match. 2: Defines the voice entity select rule as priority. 3: Defines the voice entity select rule as random select. 4: Defines the voice entity select rule as longest unused, meaning the longer the voice entity is unused, the higher its priority is. Table 1-25 Meanings of the serial numbers Serial Number Meaning 1 Exact match 2 Priority 3 Random select 4 Longest unused Description Use the select-rule rule-order command to configure the select rule of voice entities. Use the undo select-rule rule-order command to restore the default value. By default, the order of voice entity select rules is 1->2->3. That means exact match first, voice entity priority second, and random select last. Use the select-rule rule-order command to configure three rules of different priorities at most. But you cannot configure the same rules. Order of priorities is the order of select rules. If there are rules of multiple priorities, the system first select voice entities according to the rule of first priority. If that rule cannot distinguish the priorities of the voice entities, the rule of second priority will be used. The rule of third priority will be used for the voice entities that cannot be distinguished by the first and second rules, 1-95 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands and so on. If all the rules cannot distinguish voice entity priorities, Identifiers of voice entities are used for selection, and the voice entity with the smallest identifier will be preferred. The select rules are described in the following: 1) Exact match. From left to right, the more digits it matches, the better the precision is. The system stops using the rule once it meets a digit that cannot be matched uniquely. 2) Priority. Voice entities are divided into 11 classes, with values ranging from 0 to 10. The smaller the value is, the higher the priority is. That means 0 is the highest priority. 3) Random select. The system selects a voice entity from a group of qualified voice entities randomly. 4) Longest unused. The longer the voice entity is unused, the higher its priority is. There will be no collision between voice entities when random select is applied. Therefore, random select can only be used as the last rule or the only rule. Related command: select-rule search-stop and select-rule type-first. Example # Set the select rule of voice entities to exact match-> priority-> longest unused. [H3C-voice-dial] select-rule rule-order 1 2 4 1.1.84 select-rule search-stop Syntax select-rule search-stop max-number undo select-rule search-stop View Voice dial program view Parameter max-number: Maximum number of voice entities found, in the range 1 to 128. Description Use the select-rule search-stop command to configure the maximum number of voice entities found. Use the undo select-rule search-stop command to restore the maximum number of voice entities found to 128. By default, no maximum number of the search for voice entities is configured. That is, the maximum number of the search for voice entities is the default value 128. 1-96 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands There might be multiple voice entities that meet the call requirements during call connection. If a voice entity fails, the system can search for other qualified ones and continue the call. The select-rule search-stop command is used to define the maximum number of qualified voice entities to be found before the search stops. If there are multiple qualified voice entities, only the one of the highest priority is used to initiate a call. Related command: select-rule rule-order and select-rule type-first. Example # Configure to search 5 voice entities at most. [H3C-voice-dial] select-rule search-stop 5 1.1.85 select-rule type-first Syntax select-rule type-first 1st-type 2nd-type 3rd-type undo select-rule type-first View Voice dial program view Parameter 1st-type: Serial number of the type of the first priority, ranging from 1 to 3. The meanings of the serial numbers are shown in the following table. 2nd-type: Serial number of the type of the second priority, ranging from 1 to 3. It cannot be the same number as 1st-type. 3rd-type: Serial number of the type of the third priority, ranging from 1 to 3. It cannot be the same number as 1st-type or 2nd-type. Table 1-26 Meaning of the serial numbers Serial Number Meaning 1 VoIP voice entity 2 POTS voice entity 3 VoFR voice entity Description Use the select-rule type-first command to configure the type-first select rules for voice entities. Use the undo select-rule type-first command to delete the type-first select rules. 1-97 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands By default, voice entities are not selected according to their types. The command is used to configure the select order for voice entities according to their types. If multiple voice entities (of different types) are qualified for a call connection, the system will select a suitable voice entity according to the type-first rules configured by the select-rule type-first command. The order of inputting the parameters determines voice entity type priorities. The system selects the first type first and then the third type last. Related command: select-rule rule-order, select-rule search-stop. Example # Configure to select voice entities according to the type-first rules: VoIP-> POTS -> VoFR. [H3C-voice-dial] select-rule type-first 2 1 3 1.1.86 select-stop Syntax select-stop undo select-stop View Voice entity view Parameter None Description Use the select-stop command to disable the search for voice entities. Use the undo select-stop command to re-enable the search for voice entities. By default, the search for voice entities is enabled. There might be multiple qualified voice entities for a call connection. If a voice entity fails, the system can search for another one that meets the requirements and continue the call. In that case, you can use this command to configure the system to stop the search when it has found the specified voice entity. Related command: select-rule rule-order, select-rule type-first. Example # Configure the system to stop the search once voice entity 10 is found. [H3C-voice-dial-entity10] select-stop 1-98 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.87 send-busytone Syntax send-busytone { enable | time seconds } undo send-busytone { enable | time } View FXO voice subscriber-line view Parameter enable: Enables busy-tone sending on the FXO interface. time seconds: Duration of busy tone, in the range 2 to 15 seconds. It defaults to 3 seconds. This parameter is not available when busy-tone sending is disabled. Description Use the send-busytone command to enable busy tone sending on the FXO interface. Use the undo send-busytone command to disable busy tone sending on the FXO interface. This command is valid only on FXO interfaces. By default, busy tone sending is disabled. Example # Enable FXO interface 1/0/0 to send busy tone that lasts 5 seconds. [H3C-voice] subscriber-line 3/1 FXO interface encountered [H3C-voice-line3/1] send-busytone enable [H3C-voice-line3/1] send-busytone time 5 1.1.88 send-number (voice entity view) Syntax send-number { digit-number | all | truncate } undo send-number View POTS voice entity view Parameter digit-number: Number of least significant digits that are sent, ranging from 0 to 31. It is not larger than the total number of digits of the called number. 1-99 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands all: All digits of the called number. truncate: Sends the truncated called number. Description Use the send-number command to configure the number sending mode. Use the undo send-number command to restore the default mode. By default, the truncate mode is used. This command only applies to POTS voice entities. As for the numbers sent to PSTN, this command is used to control how to send the called numbers. You can not only configure to send some digits (defined by digit-number from right to left) or all digits of the called number, but also the truncated called numbers, i.e., the numbers that match the wildcard “.” at the end. Related command: dot-match and match-template. Example # Configure voice entity 10 to send the 6 least significant digits of the called number. [H3C-voice-dial-entity10] send-number 6 1.1.89 send-number (voice subscriber-line view) Syntax send-number undo send-number View FXS voice subscriber-line view Parameter None Description Use the send-number command to configure the FXS voice subscriber-line to allow the FXS to send called numbers. Use the undo send-number command to disable the FXS to send called numbers. By default, the FXS is disabled to send called numbers. Normally, the FXS interface is connected to a standard telephone, so the FXS does not need to send called numbers. But sometimes, the FXS interface is connected to the FXO interface. In this case, when the FXO receives a ring from the FXS, it imitates an off-hook and sends dial tone for the second stage of dial. With the send-number feature 1-100 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands enabled, the FXS voice subscriber-line can send called numbers to the connected FXO interface directly to free the calling party from the second stage of dial. Example # Enable the FXS to send called numbers on FXS voice subscriber-line 0/0/0. [H3C-voice-line0/0/0] send-number 1.1.90 send-ring Syntax send-ring undo send-ring View VoIP voice entity view Parameter None Description Use the send-ring command to enable the local end to play ringback tone. Use the undo send-ring command to disable the local end to play ringback tone. This configuration takes effect only in fast-connect mode. By default, the local end does not play ringback tone. Example # Enable the local end to play ringback tone. [H3C-voice-dial-entity8801] fast-connect [H3C-voice-dial-entity8801] send-ring 1.1.91 service data enable Syntax service data enable undo service data enable View Voice entity view 1-101 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter None Description Use the service data enable command to enable data call service. Use the undo service data enable command to disable data call service. By default, data call service is disabled. Example # Configure the voice entity to support data dial service. [H3C-voice-dial-entity100] service data enable 1.1.92 shutdown (voice entity view) Syntax shutdown undo shutdown View Voice entity view Parameter None Description Use the shutdown command in voice entity view to configure to change the management status of specified voice entity from UP to DOWN. Use the undo shutdown command in voice entity view to restore the voice entity management status from DOWN to UP. By default, the voice entity management status is UP. Running command shutdown will cause the voice entity unable to make calls. Example # Change the status of voice entity 4 to DOWN. [H3C-voice-dial-entity4] shutdown 1.1.93 shutdown (voice subscriber-line view) Syntax shutdown 1-102 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands undo shutdown View Voice subscriber-line view Parameter None Description Use the shutdown command in voice subscriber-line view to configure the voice subscriber-line from UP to DOWN. Use the undo shutdown command in voice subscriber-line to restore the voice subscriber-line from DOWN to UP. By default, the voice subscriber-line status is UP. This command is applicable to FXO, FXS, analog E&M subscriber-lines and digital E1 voice subscriber-line. The function of the POTS voice subscriber-line specified by the command shutdown is disabled, whereas using command undo shutdown will enable all the voice subscriber-lines on the voice card. Example # Shut down voice subscriber-line 0/0/0. [H3C-voice-line0/0/0] shutdown 1.1.94 silence-th-span Syntax silence-th-span threshold time-length undo silence-th-span View FXO subscriber-line view Parameter threshold: Silence detection threshold. If the voice signal from the switch is smaller than this value, the system regards it as silence. This threshold ranges from 0 to 200 and defaults to 20. Normally, the signal amplitude on the links without traffic is in the range 2 to 5. time-length: Duration of silence detection. Upon expiration of this duration, the system goes on-hook automatically. It ranges from 2 to 7200 (default) seconds. 1-103 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the silence-th-span command to set the silence detection parameters that have the system take automatic on-hook action. Use the undo silence-th-span command to restore the default. Silence detection-based disconnection applies to prevent an FXO interface from hanging when the busy tone parameters provided by the connected PBX are special. In normal cases, you need not to use this function to tune the busy tone detection parameters and any misconfiguration may result in incorrect disconnection. When configuring this command, you are recommended to test multiple parameter sets for the one that allows quick resource release on the FXO interface after the user hangs up while eliminating the likelihood of incorrect on-hook. Example # Set the silence detection threshold to 20 and the duration to 10 seconds. [H3C-voice-line0/0/0] silence-th-span 20 10 1.1.95 special-service Syntax special-service enable undo special-service enable View Voice dial program view Parameter enable: Enables the special-service numbers. All special-service numbers are enabled. Description Use the special-service command to enable or disable the special-service numbers. By default, the special-service numbers are disabled. The special-service numbers include: do-not-disturb, call forwarding on busy, call forwarding unconditionally, call forwarding on no reply, alarm service, and call forwarding during conversation (not supported at present). If you use the command to disable the special-service numbers, all the numbers are disabled, and vice versa. Related command: special-service, transfer-number. Example # Enable the special-service numbers. 1-104 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands [H3C-voice-dial] special-service enable 1.1.96 subscriber-line Syntax subscriber-line line-number View Any voice view except for voice entity view Parameter line-number: Subscriber line number. Description Use the subscriber-line command to enter FXS, FXO, analog E&M, or digital E1/T1 voice subscriber-line view. In the view, use subscriber-line command to enter the voice subscriber-line view. When line-number is the FXS voice subscriber-line, the system will enter the FXS voice subscriber-line view. When the line-number is the analog E&M voice subscriber-line, the system will enter analog E&M voice subscriber-line view, etc. Related command: entity. Example # Enter the view of the voice subscriber line 0/0/0 in AR46 series router voice view. [H3C-voice] subscriber-line 0/0/0 FXO interface encountered [H3C-voice-line0/0/0] 1.1.97 substitute (voice subscriber line/entity voip/entity vofr view) Syntax substitute { called | calling } list-number undo substitute { called | calling } View Voice subscriber line view, VoIP entity view, and VoFR entity view Parameter called: Applies the number substitution rules to the called number. calling: Applies the number substitution rules to the calling number. 1-105 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands list-number: Serial number of the number substitution rule list. Its value ranges from 1 to 2147483647. Description Use the substitute command to bind the PSTN calling/called number substitution rule list. Use the undo substitute command to remove the binding By default, no number substitution rule list is bound. That means the system does not perform number substitution. Firstly, you should use the number-substitute list-number command to configure the number substitution rule list in voice dial program view, VoIP entity view, or VoFR entity view and use the rule command to configure the rules in the list. Secondly, you should use the substitute command to apply the number substitution rule list to the voice subscriber line or voice entity. Note: According to network requirements, you can complete number substitution in the following two ways. z Before voice entities are matched, you can use the substitute command in the voice subscriber-line view to substitute the calling/called numbers corresponding to the specified subscriber-line. z After voice entities are matched and before a call is initiated, you can use the substitute command in voice entity view or VoFR entity view to substitute the specified calling/called numbers. Related command: number-substitute and rule. Example # Apply called number substitution rule list 6 to voice subscriber-line 0/0/0. [H3C-voice-line0/0/0] substitute called 6 1.1.98 substitute (voice dial-program/pots entity view) Syntax substitute { incoming-call | outgoing-call } { called | calling } list-number undo substitute { incoming-call | outgoing-call } { called | calling } { all | list-number } 1-106 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View Voice dial program view and POTS entity view Parameter incoming call: Apply the number substitution rules to numbers of the call coming from the network side. ougoing call: Apply the number substitution rules to number of the calls going to the network side or PSTN. called: Applies the number substitution rules to the called number. calling: Applies the number substitution rules to the calling number. list-number: Serial number of the number substitution rule list. Its value ranges from 1 to 2,147,483,647. all: Applies all number substitution rule lists. Description Use the substitute command to bind the calling/called numbers of incoming calls from the network side and outgoing calls to the network side to a number substitution rule list. Use the undo substitute incoming-call command to remove the binding. By default, no number substitution rule list is bound. That means the system does not any number substitution. Firstly, you can use the number-substitute list-number command in voice dial program view to create a number substitution rule list, and then use the rule command to configure the rules in the list. Secondly, you can use the substitute command to apply the number substitution rule list. You should follow these rules in using the command: 1) 32 number substitution rule lists can be bound at most. 2) The system searches all the rules in the number substitution rule lists in order, and it stops the search once one rule works. Note: According to network requirements, before voice entities are matched, you can use the substitute command in the voice dial program view to substitute the calling/called numbers of the calls coming from or going to the network side. Related command: number-substitute and rule. 1-107 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Example # Apply number substitution rule list 5 to the called numbers of the incoming calls from the network side. [H3C-voice-dial] substitute incoming-call called 5 # Apply number substitution rule list 6 to the called numbers of the calls going to the network side 1.1.99 terminator Syntax terminator character undo terminator View Voice dial program view Parameter character: Telephone number terminator. The valid characters are digits 0 through 9, pound sign (#), and asterisk (*). Description Use the terminator command to configure a special character string as the terminator of a telephone number whose length is variable. Use the undo terminator command to cancel the existing setting. By default, no terminator is configured. In areas where telephone numbers with variable are used, a character can be specified as a terminator in order to avoid that the system cannot dial until the dial is timeout. Related command: match-template and timer. Example # Configure using “#” as a terminator. [H3C-voice-dial] terminator # 1.1.100 timer dial-interval Syntax timer dial-interval seconds undo timer dial-interval 1-108 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View Voice subscriber-line view Parameter seconds: Maximum interval in seconds between two digits, in the range of 1 to 300 with a default of 10. Description Use the timer dial-interval command to configure the timer that the system waits for a subscriber to dial the next digit. Use the undo timer dial-interval command to restore the default time settings. This command is applied to FXO, FXS, BSV, and E1VI interfaces. This timer will restart whenever the subscriber dials a digit and will work in this way until all the digits of the number are dialed. If the timer times out before the dialing is completed, the subscriber will be prompted to hook up and the call is terminated. Setting the argument seconds of a timer to 0 will disable this timer. Example # Set the maximum duration waiting for the next digit on voice line 0/0/0 to 5 seconds. [H3C-voice-line0/0/0] timer dial-interval 5 1.1.101 timer first-dial Syntax timer first-dial seconds undo timer first-dial View Voice subscriber-line view Parameter seconds: The maximum time waiting for the first dial, which is in the range of 1 to 300 seconds, by default seconds is 10s. Description Use the timer first-dial command to configure the timer that the system waiting for a subscriber to dial the first digit. Use the undo timer first-dial command to restore the default time settings. This command is applied to FXO, FXS subscriber-lines. 1-109 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Upon the expiration of the timer, the subscriber will be prompted to hook up and the call is terminated. Setting the argument seconds of a timer to 0 will disable this timer. Example # Set the maximum duration waiting for the first dial on voice line 0/0/0 to 10 seconds. [H3C-voice-line0/0/0] timer first-dial 10 1.1.102 timer ring-back Syntax timer ring-back seconds undo timer ring-back View Voice subscriber-line view Parameter seconds: Maximum duration of ringback tone, in the range 5 to 120 seconds. It defaults to 60 seconds. This argument applies on interfaces that are analog E&M, digital E&M, FXS, or FXO. Description Use the timer ring-back command to configure the maximum duration for the system to send the ringback tone. Use the undo timer ring-back command to restore the default. This command applies only on interfaces that are analog E&M, digital E&M, FXS, or FXO. Example # Set the maximum time duration for the system to send ringback tone to eight seconds. [H3C-voice-line0/0/0] timer ring-back 8 1.1.103 timer wait-digit Syntax timer wait-digit { infinity | seconds } undo timer wait-digit 1-110 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View Voice subscriber-line view Parameter infinity: Infinite time. seconds: The maximum duration waiting for a digit, in the range 3 to 600 seconds. It defaults to 5 seconds. This argument only applies to analog and digital E&M interfaces. Description Use the timer wait-digit command to configure the maximum time duration that the system waits for a digit. Use the undo timer wait-digit command to restore the default time settings. This command is applied to analog E&M and digital E&M subscriber-lines. Setting the argument seconds of a timer to 0 will disable this timer. Example # Set the maximum duration waiting for the first dial on voice line 0/0/0 to 10 seconds. [H3C-voice-line0/0/0] timer first-dial 10 1.1.104 trace interval Syntax trace interval packets undo trace interval View Voice view Parameter packets: Number of voice packets, which is in the range of 1 to 10000 and defaults to 200. Description Use the trace interval command to configure the interval for information recording in the debugging process, that is, the number of voice packets upon the pass of which a record will be made. Use the undo trace interval command to restore the default. Example # Record debugging information for every 300 voice packets. 1-111 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands [H3C-voice] trace interval 300 1.1.105 transmit gain Syntax transmit gain value undo transmit gain View Voice subscriber-line view Parameter value: Voice output gain ranging -14.0 to 14.0 in dB with one digit after the decimal point. By default, the value is 0 dB. Description Use the transmit gain command to configure the voice subscriber-line output end gain value. Use the undo transmit gain command to restore the default value. This command is applicable to FXO, FXS, analog E&M subscriber-lines and digital E1 voice subscriber-line. When a relatively small voice signal power is needed on the output line, this command can be used to properly increase the voice output gain value to adapt to the output line signal requirement. Related command: receive gain and subscriber-line. Caution: Adjusting the gain may cause voice calls unable to be established. Therefore, when required, adjust the gain only under the direction of technical support staff. Example # Configure the voice output gain value as -6.7dB on subscriber line 0/0/0 [H3C-voice-line0/0/0] transmit gain -6.7 1-112 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.106 tunnel-on Syntax tunnel-on undo tunnel-on View VoIP voice entity view Parameter None Description Use the tunnel-on command to enable tunnel function. Use the undo tunnel-on command to disable tunnel function. By default, the tunnel function is disabled. Tunnel function can assist in negotiating process of such nonstandard H.245 message as transmitting DTMF code transparently. Only after successfully enabling the fast connect mode, can one fulfill the configuration of tunnel function. As the calling gateway, it can be decided whether or not to enable the tunnel function for each call on the router. Being the called gateway, it shall be decided whether or not to enable the tunnel function based on the status of the calling gateway. That is, if the function is enabled on calling gateway, it will also be enabled on the called gateway. Otherwise, tunnel function is disabled on both sides. In order to transmit DTMF code transparently via the fast connect mode, the tunnel function must be enabled. Otherwise DTMF code cannot be transmitted. During actual configuration, it is only necessary to fulfill this command for the VoIP voice entity at the calling gateway. Related command: fast-connect and outband. Example # Enable the tunnel function for VoIP voice entity 10. [H3C-voice-dial-entity10] fast-connect [H3C-voice-dial-entity10] tunnel-on 1.1.107 type Syntax type { 1 | 2 | 3 | 5 } undo type 1-113 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands View Voice subscriber-line view Parameter 1, 2, 3 and 5: Correspond respectively to the four signal types of analog E&M subscriber-lines, i.e. type 1, 2, 3 and 5. Description Use the type command to configure the analog E&M subscriber-line signal type. Use the undo type command to cancel the existing setting. By default, the analog E&M subscriber-line signal type is type 5. This command is only applicable to the analog E&M subscriber-line. Of the analog E&M signaling types, a PBX transmits signals on the M line (M means Mouth, indicating it is transmitted from a PBX), and receives signals from E line (E means Ear, indicating it is received by the PBX). So, as to a router that has voice function, the router receives the M signal of the PBX, and transmits E signal to the PBX. The analog E&M subscriber-line has four data transmission (signal) lines, and two or all the four lines are used according to different types: E (“Ear” or “receive”): the signal line from the router side to the PBX side. M (“Mouth” or “transmit”): the signal line from the PBX side to the router side. SG (Signal Ground): used for some types of analog E&M only. SB (Signal Battery): used for some types of analog E&M only. Example # Configure subscriber line 0/0/0 analog E&M subscriber-line type as type 3. [H3C-voice-line0/0/0] type 3 1.1.108 vad-on Syntax vad-on undo vad-on View Voice entity view Parameter None 1-114 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Description Use the vad-on command to enable silence detection function. Use the undo vad-on command to disable silence detection function. By default, the silence detection function is disabled. VAD is an abbreviation of Voice Activity Detection, generally termed as silence detection. Its basic idea is to detect and delete any silence on the basis of the difference of energy between the voice signals of people’s conversation and their silence signals, so that no signals are produced; and only when an abrupt activity tone is detected will a voice signal be generated and transmitted. It is displayed by research that no less than 50% transmission bandwidth can be attained by the use of VAD technology. Related command: entity. Example # Enable the VAD function of POTS voice entity 10. [H3C-voice-dial-entity10] vad-on 1.1.109 vi-card busy-tone-detect Syntax vi-card busy-tone-detect auto index line-number [ free | time ] View Voice view Parameter Index: Identifies busy tone detection feature types, in the range 0 to 3. line-number: Subscriber line number, the range of which is determined upon the router type and the card inserted on. It can only be an even number. free: Releases data and stop busy tone detection. time: Specifies the time for busy tone detection. Description Use the vi-card busy-tone-detect command to configure busy tone detection method on FXO subscriber-line. This command is only valid for FXO subscriber-lines, and can only be configured on the first subscriber-line of each two-subscriber-line voice board or four-subscriber-line voice board. Otherwise, the busy tone detection is not valid. The system supports four types of busy tone detection features, which are identified by the index parameter. 1-115 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands The command vi-card busy-tone-detect is applicable in most cases, which can facilitate the busy tone detection. Related command: vi-card custom-toneparam. Example # Enable busy tone detection feature 0 on the subscriber line 2. [H3C-voice] vi-card busy-tone-detect auto 2 1.1.110 vi-card cptone-custom Syntax vi-card cptone-custom type arg0 arg1 arg2 arg3 arg4 arg5 arg6 undo vi-card cptone-custom { type | all } View Voice view Parameter Type: Prompt tone type, currently supporting the following types: dial-tone, special-dial-tone, busy-tone, congestion tone, ringback-tone, and waiting-tone. arg0: Combination mode, in the range of 0 to 2. 0 indicates the overlapping of the two frequencies; 1 indicates the modulation of the two frequencies; 2 indicates the alternation of the two frequencies. arg1/arg2: Frequencies of the two single-frequency tones in Hz. The range of the two frequencies is related to the combination mode (arg0). If the mode is overlapping or alternation, the range of the two frequencies is 300 to 3400; if the mode is multiplication, the range of the two frequencies is 0 to 3400 and the absolute value of the two frequencies’ sum and difference must be in the range of 300-3400. arg3: The on duration of the first on-off ratio in ms, in the range of 30 to 8191. If the prompt tone is played continually, the value is set to 8192. arg4: The off duration of the first on-off ratio in ms, in the range of 30 to 8191. arg5: The on duration of the second on-off ratio in ms, in the range of 30 to 8191. arg6: The off duration of the second on-off ratio in ms, in the range of 30 to 8191. all: All types of prompt tones. Table 1-27 Type of prompt tone Type of prompt tone Description dial-tone Dial tone special-dial-tone Special dial tone 1-116 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Type of prompt tone Description congestion-tone Congestion tone busy-tone Busy tone ringback-tone Ringback-tone waiting-tone Waiting tone Description Use the vi-card cptone-custom command to set the prompt tone parameters of the system. Use the undo vi-card cptone-custom command to delete the prompt tone parameters. Note: After the configuration of the prompt tone parameters, these parameters cannot take effect immediately. You must use the cptone cs command in subscriber line view to validate your configuration. Example # Set the busy tone parameters, 425 Hz in single frequency, and the on or off duration is 350 ms. [Router-voice] vi-card cptone-custom busy-tone 0 425 425 350 350 350 350 1.1.111 vi-card reboot Syntax vi-card reboot slot-number View Voice view Parameter slot-number: Slot number where the voice card is located. Description Use the vi-card reboot command to reboot a voice card. 1-117 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands To reset a voice card will interrupt all the service on the voice card, and re-initialize the card status. Generally, if the voice card works abnormally (e.g. the configuration is correct but normal connection is not possible), this operation can be used. You can view the card indicator status when resetting the voice card to judge whether to perform the reset operation. First use command display version to display the distributed slots of the voice cards in the router. Related command: display version. Example # Reset the voice card of slot 3. [H3C-voice] vi-card reboot 3 1.1.112 voice-setup Syntax voice-setup undo voice-setup View System view Parameter None Description Use the voice-setup command to enter voice view and enable voice services. Use the undo voice-setup command to disable all voice services. By default, all voice services are disabled. Example # Enter voice view and enable voice services. [H3C] voice-setup Starting voice service, please waiting...... [H3C] Succeed in starting voice service! # Disable all voice services. [H3C] undo voice-setup Warning: Voice service will be stopped! Continue? [Y/N]y Stopping voice service, please waiting...... 1-118 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands [H3C] Succeed in stopping voice service! 1.1.113 voip calledtunnel Syntax voip calledtunnel enable undo voip calledtunnel enable View Voice view Parameter enable: Enables the tunnel function on the called gateway. Description Use the voip calledtunnel enable command to enable the tunnel function on the called gateway. Use the undo voip calledtunnel enable command to disable the tunnel function on the called gateway. By default, the tunnel function on the called gateway is enabled. When the router is a called gateway, it decides whether to use the tunnel function depending on the configuration of the voip calledtunnel command. To interoperate with devices that do not support tunneling using H.323 recommendation, you need to disable with the undo form of this command the tunnel function on the called gateway. Related command: tunnel-on. Example # Disable the tunnel function on the called gateway. [H3C-voice] undo voip calledtunnel enable 1.1.114 voip call-start Syntax voip call-start { fast | normal } undo voip call-start View Voice view 1-119 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter fast: The called GW initializes calls in a fast way. normal: The called GW initializes calls in a non-fast way. Description Use the voip call-start command to configure a call initialization mode for the called GW. Use the undo voip call-start command to restore the default. By default, the fast mode is used. As the process of faculty negotiation is omitted in fast connect procedures, the faculties of the two parties are determined by the GW. If a router acts as a calling GW, you can enable or disable fast connect for each channel of initiated calls. If it acts as a called GW, it will use or not use the fast connect mode to initialize calls depending on the parameters of the voip call-start command, in the case that the calling GW uses the fast connect mode. For related configurations, see fast-connect command. Example # Configure the called gateway to initialize calls in normal mode. [H3C-voice] voip call-start normal 1.1.115 voip h323-descriptor Syntax voip h323-descriptor descriptor undo voip h323-descriptor View Voice view Parameter descriptor: The description character string of H.323, in the length of 1 to 64. By default, the descriptor is "Comware Gateway". Description Use the voip h323-descriptor command to configure the description character string of the voice gateway H.323. It is recommended to configure with default value, also you can configure according to the actual requirements. If both ends are the device based on Comware, and the 1-120 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands attribute is configured at both ends (i.e., the default value is not adopted), the character strings of both ends must be the same. Example # Configure the descriptor of H.323 as “mystring”. [H3C-voice] voip h323-descriptor mystring 1.1.116 voip timer Syntax voip timer voip-to-pots seconds undo voip timer voip-to-pots View Voice view Parameter voip-to-pots: Delays waiting for switching from VoIP entity to backup POTS entity after a VoIP call failure. seconds: Time length in seconds, in the range of 3 to 30 with a default of 5. Description Use the voip timer command to set the VoIP-to-POTS timer, or the delay waiting for switching from VoIP entity to backup POTS entity after a VoIP call is failed. Use the undo voip timer command to restore the default. Example # Configure the delay waiting for switching from the VoIP entity to the backup POTS entity after a call failure to 10 seconds. [H3C] voip timer voip-to-pots 10 1.1.117 vqa data-statistic Syntax vqa data-statistic enable undo vqa data-statistic enable View Voice view 1-121 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Parameter enable: Enables counting of voice data. Description Use the vqa data-statistic command to enable counting of voice data. Use the undo vqa data-statistic enable command to disable counting of voice data. By default, counting of voice data is disabled. In order to quickly locate trouble in VoIP call and perform debugging, use the vqa data-statistic enable command to enable statistics of voice data. Information to be performed statistics includes time for successfully searching the voice table, total number of received data packets, time for searching the table in fast and common modes and voice and fax of receive and transmit channels. Statistics of voice data mainly serves debugging. Therefore, the user is recommended to disable this function when the service is normal so as to ensure higher performance of voice data processing. Related command: reset voice voip data-statistic, display voice voip data-statistic. Example # Enable counting of voice data. [H3C-voice] vqa data-statistic enable 1.1.118 vqa dscp Syntax vqa dscp { media | signal } dscp-value undo vqa dscp { media | signal } View Voice view Parameter media: Global DSCP value in the ToS field of the IP packets that carry RTP streams. signal: Global DSCP value in the ToS field of the IP packets that carry voice signaling. dscp-value: DSCP value in the range 0 to 63 or the keyword ef, af11, af12, af13, af21, af22, af23, af31, af32, af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, or zero. 1-122 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Table 1-28 DSCP values Keyword DSCP value in binary DSCP value in decimal ef 101110 46 af11 001010 10 af12 001100 12 af13 001110 14 af21 010010 18 af22 010100 20 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 zero 000000 0 Description Use the vqa dscp command to set the global DSCP value in the ToS field in the IP packets that carry the RTP stream or voice signaling. Use the undo dscp media command to restore the default DSCP, that is, zero (00000). 1-123 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands Note: The function of this command is the same as the command used for setting DSCP in the “QoS” part of this manual. If two DSCP values are configured, the one configured in the “QoS” part takes priority. This command is valid for both SIP and H.323. Example # Set the DSCP value in the ToS field in the IP packets that carry voice signaling to af41. [H3C] voice-setup [H3C-voice] vqa dscp signal af41 1.1.119 vqa dsp-monitor Syntax vqa dsp-monitor buffer-time time undo vqa dsp-monitor buffer-time View Voice view Parameter buffer-time time: Duration of DSP buffer data monitoring, in the range of 180 to 480 ms. Description Use the vqa dsp-monitor command to set duration for DSP buffer data monitoring. Use the undo vqa dsp-monitor buffer-time command to remove the setting. By default, DSP data is not monitored. If the DSP buffering duration exceeds the preset threshold, DSP will discard voice data to keep the voice delay with a rational range. A time value greater than 240 is recommended for too small duration value will result in poor voice quality in the case of severe jitter. Related command: ip-precedence. Example # Set the duration for DSP buffer data monitoring to 270 milliseconds. [H3C-voice] vqa dsp-monitor buffer 270 1-124 Command Manual – Voice Comware V3 Chapter 1 VoIP Configuration Commands 1.1.120 vqa performance Syntax vqa performance { send | receive } [ fast | normal ] undo vqa performance { send | receive } View Voice view Parameter send: Configures voice data sending process. receive: Configures voice data receiving process. fast: Configures the sending or receiving process of voice data to fast forwarding process. normal: Configures the sending or receiving process of voice data to normal forwarding process. Description Use the vqa performance command to configure the sending or receiving process of voice data to fast or normal forwarding process. Use the undo vqa performance command to restore the sending and receiving process of voice data to normal forwarding process. By default, the sending and receiving process of voice data is normal. Example # Configure the current sending process of voice data to fast forwarding process. [H3C-voice] vqa performance send fast The configuration works immediately! 1-125 Command Manual – Voice Comware V3 Chapter 2 BSV Configuration Commands Chapter 2 BSV Configuration Commands 2.1 BSV Configuration Commands 2.1.1 permanent-active Syntax permanent-active undo permanent-active View BSV interface view (in NT mode) Parameter None Description Use the permanent-active command to configure BSV interfaces to permanent active. Use the undo permanent-active command to cancel the setting. By default, BSV interfaces are not permanent active (undo permanent-active). Example # Enable the BSV interfaces. [H3C-Bsv2/1/0] permanent-active 2.1.2 power-source Syntax power-source undo power-source View BSV interface view (in NT mode) Parameter None 2-1 Command Manual – Voice Comware V3 Chapter 2 BSV Configuration Commands Description Use the power-source command to enable remote power supply. Use the undo power-source command to cancel the setting. By default, remote power supply is disabled. Note: In network side mode, BSV V2 circuit board can provide remote power supply of 4w at maximum. Example # Configure to enable remote power supply on BSV interfaces. [H3C-bsv2/1/0] power-source Note: Other ISDN commands available with BSV interfaces include: isdn bch-local-manage, isdn bch-select-way, isdn caller-number, isdn calling, isdn check-called-number, isdn ignore connect-ack, isdn ignore hlc, isdn ignore llc, isdn ignore sending-complete, isdn l3-timer, isdn link-mode, isdn number-property, isdn overlap-sending, isdn protocol-mode, isdn protocol-type, isdn statistics, and isdn two-tei. For detailed information, refer to Chapter 3 “ISDN Configuration Commands” of Link Layer Protocol Command Manual. 2-2 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Chapter 3 VoFR Configuration Commands 3.1 VoFR Configuration Commands 3.1.1 address Syntax address { vofr-dynamic serial interface-number dlci-number | vofr-static serial interface-number dlci-number cid-number | x121 x121-address } undo address { vofr-dynamic | vofr-static | x121 } View Voice entity view Parameter vofr-dynamic serial interface-number: Session target interface of a VoFR voice entity in dynamic mode. vofr-static serial interface-number: Session target interface of a VoFR voice entity in static mode. dlci-number: Session target DLCI of a VoFR voice entity, in the range 16 to 1007. cid-number: FRF.11 subchannel number of the VoRF entity, in the range of 4 to 256. x121 x121-address: X.121 address of destination host. By default, the X.121 address is null. Description Use the address command to configure a route to the peer voice gateway or assign an X.121 address for a destination host. Use the undo address command to cancel the setting. By default, no route to the peer voice gateway is configured. The X.121 address configured in this command must be the X.121 address used by the destination host. Configuring the address of VoFR voice entity in vofr-dynamic mode includes configuring output interface and outgoing DLCI number. Configuring the session target of VoFR voice entity in vofr-static mode includes configuring output interface, outgoing DLCI number, FRF.11 subchannel number and creating voice channel without delay. Voice channel is deleted when deleting the address. Related command: match-template, call-mode, and vofr. 3-1 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Example # Set the X.121 address of the destination host to 10001. [H3C-voice-dial-entity4] address x121 10001 # Assign 100 to the DLCI for the session target Serial 1/0/0 corresponding to the called number 12345. [H3C-voice-dial-entity4] match-template 12345 [H3C-voice-dial-entity4] address vofr-dynamic serial1/0/0 100 # Set DLCI to 100 and CID to 6 for the session target Serial 1/0/1 in vofr-static mode. 3.1.2 call-mode Syntax call-mode { static | dynamic } undo call-mode [ static | dynamic ] View Voice entity view Parameter static: FRF.11 trunk mode is adopted. dynamic: Dynamic calling is adopted. Description Use the call-mode command to configure a call-mode for calls between local router and peer router. Use the undo call-mode command to restore the default value. By default, the dynamic mode is adopted. If dynamic is configured in this command, which call signaling should be adopted will be specified by the vofr command on the DLCI. Note: If frf11-trunk is configured in this command, the trunk-id command must be configured for the VoFR voice entity of the called party. Related command: trunk-id, address. 3-2 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Example # Configure static mode for the VoFR voice entity 10. [H3C-voice-dial-entity10] call-mode static 3.1.3 cid select-mode Syntax cid select-mode { max-poll | min-poll } undo cid select-mode View DLCI view Parameter max-poll: Cyclically selects routes in descending order. min-poll: Cyclically selects routes in ascending order. Description Use the cid select-mode command to configure the route selection mode used by the calling party to set up VoFR calls. Use the undo cid select-mode command to restore the default. By default, max-poll mode applies. When multiple channels of voice are multiplexed onto the same DLCI, using the same voice CID at both ends can cause call collision. To prevent this from happening and to avoid call loss, you may configure different route selection modes on the two ends. Related command: vofr. Example # Adopt min-poll route selection mode on DLCI 100. [H3C-Serial1/0/0] fr dlci 100 [H3C-fr-dlci-Serial1/0/0-100] vofr h3c-switch [H3C-fr-dlci-Serial1/0/0-100] cid select-mode min-poll 3.1.4 debugging voice vofr Syntax debugging voice vofr { all | error | fax | intf | motorola | nonstandard | timer | vcc | vofr | vpp } 3-3 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands View User view Parameter all: Enables all VoFR module debugging. error: Enables debugging of VoFR module errors resulting in connection failures. fax: Enables debugging of information between the VoFR and Fax modules. intf: Enables debugging of information between the VoFR and FR modules. motorola: Enables debugging of motorola signaling packets in the VoFR module. nonstandard: Enables debugging of nonstandard signaling packets in the VoFR module. timer: Enables debugging of the operations involving timer in the VoFR module. vcc: Enables debugging of information between VoFR and VCC modules. vofr: Enables debugging of information between the VoFR module peers. vpp: Enables debugging of information between the VoFR module and the lower layer VPP module. Description Use the debugging voice vofr command to enable VoFR module debugging. Example # Enables debugging of VoFR module errors resulting in connection failures. debugging voice vofr vpp error 3.1.5 display fr vofr-info Syntax display fr vofr-info [ serial interface-number [ dlci-number ] ] View Any view Parameter serial interface-number: Interface needs to be displayed. dlci-number: Number of the DLCI that needs to be displayed, in the range 16 to 1007. Description Use the display fr vofr-info command to view the VoFR information. 3-4 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands This command is used to display the use of FRF.11 sub-channels on a VoFR DLCI. It supports three displaying types: z Neither interface nor DLCI is specified z Specifies interface, but not DLCI z Specifies both interface and DLCI Example # Specify the interface and DLCI on which the Frame Relay VoFR information will be displayed. [H3C] display fr vofr-info INTERFACE(DLCI) VOFR-MODE CID CID-TYPE Serial1/0/0(200) VOFR-H3C 4 DATA Serial1/0/0(200) VOFR-H3C 5 FRAG-DATA Table 3-1 Description on the fields of the display fr vofr-info command Field Description INTERFACE Interface unit DLCI PVC identifier VOFR-MODE Type of VoFR CID Calling ID CID_TYPE Type of calling ID 3.1.6 display voice vofr call Syntax display voice vofr call [ channel ] View Any view Parameter channel: VPU channel number. Description Use the display voice vofr call command to view information of the current call. This command is used to display the information of incoming and outgoing call control blocks, the connection state of the modules, call state, calling number, called number, etc. 3-5 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Example # Display the information of the current call. [H3C] display voice vofr call CCB VCC CRV: 0 VOFR CRV: 0 Call Protocol: frf11 trunk Status: idle Call Mode: idle Call Attrib: normal Release Type: Fr Release Type: Timer: 0 FREX CCXID: 16181204 VPU Channel: Dialpeer Line Number: 103 Voice Bandwidth: 0 Packet Size: 30 Encode Type: G729R8 Decode Type: G729R8 DTMF-Relay: disable Voice Vad Mode: no Use Sequence: no Use TimeStamp: no Caller Number: Called Number: 102 Voice Packet Time: 0 Run Mode: idle Fax Is Running: no Max Fax Transmit rate: voice Fax Start Mode: passive Fax trainning result: 0 Fax Send NSF: no Switch Timer: 0 ID of Fax Start Tone: 0 Threshold of TCF error: 10 Fax Tx carrier power in db: 15 Packet number in Fax data buffer: 0 Fax Release Reason: Fax Bandwidth: 0 3-6 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Table 3-2 Description of VoFR call statistics information Field Description CCB VCC CRV Call reference value of VCC module VOFR CRV Call reference value of VoFR module Call Protocol Protocol type of VoFR presently Status Status of present call Call Mode Mode of the call, such as calling, called and idle Call Attrib Attribute of the call, such as normal or busy Release Type Cause of releasing the VoFR call Fr Release Type Cause of releasing the FR connection Timer Flag of timer of the call FREX CCXID Call decryption identifier at interface of FREX VPU Channel Channel of VPU module Dialpeer Port Number Number of voice entity of the call Voice Bandwidth Voice bandwidth of the call Packet Size Packets length of the call Encode Type Compression type of the call Decode Type Decompression type of the call DTMF-Relay Out-of-band or in-band call transmission is enabled or disabled. Voice Vad Mode Voice Activity Detection is used or not Use Sequence Sequence is used or not for the call Use TimeStamp Timestamp is used or not for the call Caller Number Calling number of the call Called Number Called number of the call Voice Packet Time Time of voice packets Run Mode Call mode of gateway currently, such as voice and realtime fax. Fax Is Running Fax communication is running or not Max Fax Transmit rate Max transmit rate of fax communication Fax Start Mode Start mode of fax, such as active or passive Fax training result Train result of fax communication Fax Send NSF Negotiation mode of fax faculty 3-7 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Field Description Switch Timer Flag of switching timer ID of Fax Start Tone Identifier of saving fax start tone Threshold of TCF error Local-train threshold of fax at local Fax Tx carrier power in db Energy level of gateway carrier transmitting Packet number in Fax data buffer Number of fax packets in buffer Fax Release Reason Cause of releasing fax connection Fax Bandwidth Bandwidth of fax 3.1.7 display voice vofr statistic Syntax display voice vofr statistic { all | call | error | timer | vcc | vofr | vpp } View Any view Parameter all: Displays all the statistics of VoFR module. call: Displays the call statistics in VoFR module. error: Displays the statistics of VoFR module errors resulting in connection failures. timer: Displays statistics of operations involving timer in the VoFR module. vcc: Displays the statistics of the interactive messages between VoFR and VCC modules. vofr: Displays the statistics of information between the VoFR module peers. vpp: Displays the statistics about the information between VoFR module and the lower layer VPP module. Description Use the display voice vofr statistic command to view the call statistics between the VoFR module and VCC, VPP and other modules. Example # Display the call statistics between the VoFR module and other modules. [H3C] display voice vofr statistic call Statistic about VOFR call : { 3-8 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Call number as Caller: 0 Succeed call number as Caller: 0 Fail call number as Caller: 0 Call number as Called: 0 Succeed call number as Called: 0 Fail call number as Called: 0 Send voice packets number: 0 Receive voice packets number: 0 Lost voice packets received number: 0 } Table 3-3 Description of call statistics between the VoFR module and other modules Field Description Call number as Caller Times of calling out Succeed call number as Caller Times of calling out correctly Fail call number as Caller Times of calling out wrongly Call number as Called Times of calling in Succeed as number from Called Times of calling in correctly Fail call number as Caller Times of calling in wrongly Send voice packets number Number of packets sent Receive voice packets number Number of packets received Lost voice packets received number Number of packets dropped 3.1.8 motorola base-svc Syntax motorola base-svc start-svc undo motorola base-svc View DLCI view Parameter start-svc: Start SVC for voice routing, in the range 1 to 4080. By default, the start SVC for voice routing is 17. 3-9 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Description Use the motorola base-svc command to configure the start SVC for voice routing on a DLCI. Use the undo motorola base-svc command to restore the default. The user is allowed to configure this command when the VoFR call control protocol on the DLCI is Motorola-compatible. Related command: vofr, motorola max-voice. Example # Specify the start SVC for voice routing on DLCI 100 to 100. [H3C-fr-dlci-Serial1/0/0-100] motorola base-svc 100 3.1.9 motorola encapsulation Syntax motorola encapsulation { codex | rfc1294 } undo motorola max-voice View DLCI view Parameter codex: Motorola codex encapsulation. rfc1294: RFC 1294 multi-protocol encapsulation. Description Use the motorola encapsulation command to specify an encapsulation for Motorola LCON on the DLCI. Use the undo motorola encapsulation command to restore the default. By default, codex encapsulation mode is used. This command is available only when the VoFR call control protocol is set to motorola-compatible on the DLCI. Related command: vofr, motorola max-voice. Example # Adopt RFC 1294 multi-protocol encapsulation for Motorola LCON on the DLCI 200. [H3C-fr-dlci-Serial1/0/0-200] motorola encapsulation rfc1294 3-10 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands 3.1.10 motorola max-voice Syntax motorola max-voice max-number undo motorola max-voice View DLCI view Parameter max-number: The maximum number of voice channels, in the range 0 to 15. By default, the maximum number of voice channels is 0. Description Use the motorola max-voice command to configure the maximum number of voice channels created on a DLCI concurrently. Use the undo motorola max-voice command to restore the default. The user can configure this command only when the VoFR call control protocol used on the DLCI is motorola-compatible. If it is set to 0, no voice will be sent. Related command: vofr. Example # Specify that a maximum of 10 voice channels can be concurrently set up on DLCI 100. [H3C-fr-dlci-Serial1/0/0-100] motorola max-voice 10 3.1.11 motorola remote-id Syntax motorola remote-id index undo motorola remote-id View DLCI view Parameter index: Remote connection ID, in the range 1 to 255. It defaults to 1. Description Use the motorola remote-id command to configure a remote connection ID. 3-11 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Use the undo motorola remote-id command to restore the default. Example # Set the remote connection ID to 20. [H3C-fr-dlci-Serial1/0/0-200] motorola remote-id 20 3.1.12 outband vofr Syntax outband vofr undo outband View Voice entity view Parameter None Description Use the outband vofr command to configure out-of-band transmission of DTMF codes. Use the undo outband command to restore transmission of DTMF codes to in-band mode. By default, in-band transmission is adopted. In practice, to implement the transparent transmission of DTMF codes, the transmission mode of DTMF codes should be configured in the VoFR voice entity on the calling gateway as well as the POTS voice entity on the called gateway. Related command: compression. Example # Configure VoFR voice entity 10 to transmit DTMF codes out of band. [H3C-voice-dial-entity10] outband vofr 3.1.13 priority Syntax priority priority-number undo priority 3-12 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands View Voice entity view Parameter priority-number: Voice entity priority in the range 0 to 10, with 0 being the highest priority and 10 being the lowest. The default is 0. Description Use the priority command to assign a priority to the voice entity. Use the undo priority command to restore the default. When the calling party places a call, it will dial by the voice entity reference in the case that multiple voice entities are found for the called number. The voice entity with the highest preference will be used first. Related command: entity. Example # Set the priority of VoFR voice entity 2001 to 5. [H3C-voice-dial-entity2001] priority 5 3.1.14 send-called number Syntax send-called-number undo send-called-number View VoFR voice entity view Parameter None Description Use the send-called-number command to enable the local end to send the called number when originating a call. Use the undo send-called-number command to disable the local end from sending the called number when originating a call. By default, the local end does not send the called number when originating a call. 3-13 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Note: The send-called-number command is valid only when the VoFR dynamic call control protocol is Motorola-compatible. Example Enable VoFR voice entity 2000 to send the called number when a local call is originated. [H3C-voice-dial-entity2000] send-called-number 3.1.15 seq-number Syntax seq-number undo seq-number View Voice entity view Parameter None Description Use the seq-number command to enable voice packets transmitted on the local gateway to carry sequence numbers. Use the undo seq-number command to disable voice packets to carry sequence numbers. By default, no sequence numbers are carried in voice packets. The gateway at receiving end can know whether the voice packets have been dropped, repeated or disordered according to the numbers, so that voice compensation can be made. However, use of numbers requires more bandwidth. You must carefully consider the trade-off between packet numbering and bandwidth. Related command: compression. 3-14 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Note: The configuration of this command will not affect CODEC numbering. If needed, DSP will number CODEC, regardless of whether this command has been configured. Example # Configure that the voice packets should be numbered in the VoFR voice entity 10. [H3C-voice-dial-entity10] seq-number 3.1.16 timestamp Syntax timestamp undo timestamp View VoFR voice entity view Parameter None Description Use the timestamp command to enable the local end to send voice packets with timestamps. Use the undo timestamp command to disable voice packets to carry timestamps. By default, voice packets do not carry timestamps. Example # Enable voice packets to carry timestamps in VoFR voice entity 10. [H3C-voice-dial-entity10] timestamp 3.1.17 trunk-id Syntax trunk-id string undo trunk-id View Voice entity view 3-15 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Parameter string: The number that the local end dials to call the PSTN side in FRF.11 trunk mode. It is a digit string of 1 to 31 at length. Description Use the trunk-id command to configure the number that the local end dials to call the PSTN side in FRF.11 trunk mode. Use the undo trunk-id command to restore the local called number to null. By default, no number is configured, that is means the parameter string is null. This command can be used only for the VoFR voice entities in FRF.11 trunk mode. FRF.11 does not provide end-to-end signaling management leased line. To create a leased line, the trunk-id command must be configured. Related command: match-template, call-mode. Example # Set the outgoing number to PSTN in VoFR voice entity 2001 to 2001. [H3C-voice-dial-entity1] trunk-id 2001 3.1.18 vad-on Syntax vad-on undo vad-on View Voice entity view Parameter None Description Use the vad-on command to enable voice activity detection (VAD). Use the undo vad-on command to disable VAD. By default, the silence detection function is disabled. VAD is also known as silence detection. Its basic idea is to detect and delete any silence on the basis of the difference of energy between the voice signals of people’s conversation and their silence signals, so that no signals are produced; and only when abrupt activity tone is detected will a voice signal be generated and transmitted. The 3-16 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands research showed that at least 50 percent of transmission bandwidth can be attained by the use of VAD technology. Related command: entity. Example # Enable VAD for VoFR voice entity 2. [H3C-voice-dial-entity2] vad-on 3.1.19 vofr Syntax vofr { h3c-switch [ dte | dce ] | motorola-compatible [ dte | dce ] | nonstandard-compatible signal-channel ccid-no data-channel dcid-no [ keepalive ] } undo vofr View DLCI view Parameter signal-channel ccid-no data-channel dcid-no: Indicates that VoFR works in nonstandard-compatible mode and specifies FRF.11 sub-channels for signaling and data. The subchannel is in the range 4 to 255. keepalive: Indicates whether the keepalive message will be regularly sent. In Non-standard-compatible mode, the keepalive message is regularly sent to monitor the control subchannel status. If the keepalive argument is configured, one end will assume that congestion has occurred to the network if no keepalive message is received for a certain period. The active call control subchannel will be deactivated, and thus voice call can no longer be set up. If the argument is not configured, the status of the control subchannel is synchronous to the PVC status. h3c-switch: Adopts H3C private mode. motorola-compatible: Adopts Motorola-compatible mode for compatibility with VoFR of Motorola routers. nonstandard-compatible: Adopts nonstandard-compatible mode for compatibility with VoFR of Cisco routers. dte: Annex G control block works as DTE. dce: Annex G control block works as DCE. Description Use the vofr command to enable VoFR operation mode for a DLCI. 3-17 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Use the undo vofr command to disable VoFR operation mode for the DLCI. By default, VoFR is not supported. If call-mode static has been configured to use FRF.11 trunk mode, configuring Motorola-compatible mode in this command will cause session failure. In Motorola-compatible mode, a DLCI cannot carry both dynamic calling and FRF.11 trunk. In H3C-switch or Motorola-compatible mode, use the T1.167 Annex G protocol. When making configuration, set different ANNEX G control block types at both ends to different modes, that is, one end to DTE and the other end to DCE. Related command: call-mode. Example # Specify voice calling on DLCI 100 to nonstandard-compatible, call control signaling subchannel “ccid” to 4, data subchannel dcid to 5 and keepalive messages to be sent timely. [H3C-fr-dlci-Serial1/0/0-100] vofr Nonstandard-compatible signal-channel 5 data 4 keepalive # Specify voice call mode on dlci 200 to H3C-switch (DTE). [H3C-fr-dlci-Serial1/0/0-100] vofr h3c-switch # Specify voice call mode on dlci 300 to Motorola-compatible (DCE). [H3C-fr-dlci-Serial1/0/0-100] vofr motorola-compatible dce 3.1.20 vofr frf11-timer Syntax vofr frf11-timer time View Voice view Parameter time: Interval of Trunk Wait timer, in the range 10 to 600 seconds. The default interval is 30 seconds. Description Use the vofr frf11-timer command to configure the interval of Trunk Wait timer in FRF.11 trunk mode. All the local sessions in FRF.11 trunk mode use the same timer. Related command: call-mode 3-18 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Example # Set the period of the Trunk Wait timer in FRF.11 trunk mode to 40 seconds. [H3C-voice] vofr frf11-timer 40 3.1.21 vofr jitter-buffer Syntax vofr jitter-buffer length View Voice view Parameter length: Jitter buffer length in the range 0 to 8. If it is set to 0, jitter buffer will not be used. The default jitter buffer length is 0. Description Use the vofr jitter-buffer command to configure a jitter buffer length. If delay jitter is serious in voice transmission, setting jitter buffer can smooth such jitter. However, this undertaking will prolong voice delay. Therefore, the user should carefully consider the trade-off between delay and delay jitter when setting jitter buffer length. Example # Set the length of jitter buffer to 5. [H3C-voice] vofr jitter-buffer 5 3.1.22 voice bandwidth Syntax voice bandwidth reserved-bps [ reserved ] undo voice bandwidth View Frame relay class view Parameter reserved-bps: Maximum bandwidth for voice calls, in the range 8000 bps to 45,000,000 bps. reserved: Reserves bandwidth for a call until it is terminated. 3-19 Command Manual – Voice Comware V3 Chapter 3 VoFR Configuration Commands Description Use the voice bandwidth command to configure maximum bandwidth for voice calls. Use the undo voice bandwidth command to restore the default. By default, no reserved bandwidth is configured for voice calls. Example # Set maximum bandwidth to 200,000 bps and enable bandwidth reservation for voice calls. [H3C-fr-class-1] voice bandwidth 200000 reserved 3-20 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Chapter 4 E1/T1 Voice Configuration Commands 4.1 E1/T1 Voice Configuration Commands 4.1.1 ani Syntax ani { all | ka } undo ani View R2 CAS view Parameter all: Requires the remote end to send the category of the calling party and calling number. ka: Requires the remote end to send the category of the calling party only. Description Use the ani command to enable or disable the terminating point to send the calling party information (service category and calling number) to the originating point during call connecting process. Use the undo ani command to cancel the above configuration. By default, the terminating point does not send the calling party information to the originating point. Note: z Configure the local end with this command to support automatic number identification. z This command applies to E1 voice only. z Normally the all keyword is configured. To prevent call failures, use the ka keyword only when required by the connected switch. Related command: cas and ani-offset. 4-1 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example # Configure that the local exchange requests the opposite exchange to send the category of the calling party and calling number during the connecting process. [H3C-cas1/0/0:0] ani all 4.1.2 ani-offset Syntax ani-offset number undo ani-offset View R2 CAS view Parameter number: The number quantity collected, with a value range being the integer from 1 to10. By default, the value of number is 1. Description Use the ani-offset command to configure the number of digits of the called number to be collected prior to requesting the calling party information. Use the undo ani-offset command to restore the default value. This command applies to E1 voice only. This command is used for setting the number of digits of the called number to be collected prior to requesting the caller number or caller identifier. When the quantity of the collected numbers is less than this value, the system will wait to the next number till timeout, and in the waiting it will not request the caller number information from the remote end. When the quantity of the collected numbers is equal to or exceeds this value, it is able to request the caller number or caller identifier from the remote end. Related command: cas, timer, reverse, renew. Example # Set to start requesting the caller number or caller identifier after receiving the 3-digit numbers. [H3C-cas1/0/0:0] ani-offset 3 4.1.3 answer Syntax answer { enable | disable } 4-2 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands View R2 CAS view Parameter enable: The originating points requires that the terminating point must send answer signal. Only after the originating point receives the signal, can the peers enter the communicating state. disable: The originating point does not require the terminating point to send answer signal. In case the terminating point does not send answer signal, the originating point automatically report the answer message to the upper layer when the timer times out to have the peers enter the communicating state. Description Use the answer command to configure whether the terminating point is required to send answer signal. By default, the terminating point is required to send answer signal. This command applies to E1 voice only. In some countries, answer signals are not sent in the R2 line signaling coding scheme. In this case, you can use the answer command to adjust this signaling coding scheme. If the originating point does not require answer signals, the terminating point confirms call connection after the specified time. Related command: re-answer, timer dl re-answer. Example # Disable the terminating point to send answer signals. [H3C-cas1/0/0:0] answer disable 4.1.4 cas Syntax cas ts-set-number View CE1/PRI interface view Parameter ts-set-number: The serial number of the Timeslot set predefined, with a value range being the integer from 0 to 30. 4-3 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Description Use the cas command to enter the R2 CAS view and digital E&M signaling view. The command cas is used to enter the custom R2 CAS view. Under this mode various parameters of the R2 signaling in the E1 interface can be configured as required. To validate the custom R2 parameters, it is required to keep the parameter of ts-set-number in the command cas consistent with the parameter set-number in the command timeslots-set. Related command: timeslots-set, ani-offset, effect-time, reverse, select-mode, timer, trunk-direction and renew. Example # Enter the R2 CAS view of No.5 Timeslot set. [H3C-E1 1/0/0] timeslot-set 5 timeslot-list 1-31 signal r2 [H3C-E1 1/0/0] cas 5 4.1.5 clear-forward-ack Syntax clear-forward-ack { enable | disable } View R2 CAS view Parameter enable: Enables the terminating point to acknowledge the clear-forward signal by sending a clear-back signal. disable: Disables the terminating point to acknowledge the clear-forward signal by sending a clear-back signal. Description Use the clear-forward-ack command to enable or disable the terminating point to respond by sending a clear-back signal when the originating point (calling party) disconnects a call. By default, the terminating point does not send a clear-back signal to acknowledge the clear-forward signal. This command applies to E1 voice only. In some countries, if the terminating point controls the relay circuit reset in the R2 signaling exchange process, when the calling party disconnects a call and the originating point sends a clear-forward signal to the terminating point, the terminating 4-4 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands point sends a clear-back signal as an acknowledgement, and then sends a release guard signal to indicate that the line of the terminating point is thoroughly released. Related command: mode china default-standard. Example # Enable the terminating point to send a clear-back signal to acknowledge the clear-forward signal. [H3C-cas1/0/0:0] clear-forward-ack enable 4.1.6 debugging voice r2 Syntax debugging voice r2 { all | ccb controller slot-number timeslots-list | dl | dtmf | error | mfc | msg | rcv | warning } View User view Parameter all: Enables all the debugging in R2 software module. ccb controller: Enables all the CCB debugging in R2 signaling. dl: Enables the line signaling debugging in R2 signaling. dtmf: Enables the Dual-Tone Multifrequency (DTMF) debugging in R2 signaling. error: Enables the error debugging in R2 software module. msg: Enables the message interface debugging in R2 signaling. mfc: Enables the interregister signaling debugging in R2 signaling. rcv: Enables the RCV software module debugging in R2 software module. warning: Enables warning debugging in R2 software module. slot-number: E1 port number. timeslots: Time slot number, ranging from 1 to 31. Description Use the debugging voice r2 command to enable the corresponding debugging in R2 signaling module. Use the debugging voice r2 ccb command to view the information of the corresponding control block by specifying the E1 port number and time slot number. This command applies to E1 voice only. 4-5 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example # Enable all the line signaling debugging in R2 signaling on E1 voice port. debugging voice r2 dl 4.1.7 debugging voice rcv r2 Syntax debugging voice rcv r2 View User view Parameter r2: Enables the debugging between the RCV module and the R2 module of bottom layer. Description Use the debugging voice rcv r2 command to enable the debugging between the RCV module and the R2 module of bottom layer. This command applies to E1 voice only. Example None 4.1.8 debugging voice vpp r2 Syntax debugging voice vpp r2 View User view Parameter r2: Enables the debugging between the VPP module and the R2 module of bottom layer. Description Use the debugging voice vpp r2 command to enable the debugging between the VPP module and the R2 module of bottom layer. This command applies to E1 voice only. 4-6 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example None 4.1.9 default Syntax default View R2 CAS view Parameter None Description Use the default command to restore the default of all the R2 configurations and reset the R2 call statistics to 0. The following table gives the configuration items affected by this command. Table 4-1 Configuration items affected by the default command Configuration item The originating exchange acknowledgement signal. sends Default the seizure ENABLE The originating exchange sends the answer signal. ENABLE The originating exchange answers at the time of clear-back. DISABLE The originating exchange answers with the clear-back signal at the time of clear-forward. DISABLE Answer signal timeout interval 60 seconds Clear-forward signal timeout interval 10 seconds Seizure signal timeout interval 1 second Re-answer timeout interval 1 second Delay before sending the release-guard signal upon a timeout 10 second Seizure acknowledgement signal timeout interval 40 seconds The ABCD bit pattern that represents an idle receive line 1001 The ABCD bit pattern that represents an idle transmit line 1001 4-7 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Configuration item Default The ABCD bit pattern that represents a seized receive line 0001 The ABCD bit pattern that represents a seized transmit line 0001 The ABCD bit pattern that represents the seizure acknowledged state of the transmit line. 1101 The ABCD bit pattern that represents the seizure acknowledged state of the receive line. 1101 The ABCD bit pattern that represents the answered state of the receive line. 0101 The ABCD bit pattern that represents the answered state of the transmit line. 0101 The ABCD bit pattern that represents the clear-forward state of the receive line. 1001 The ABCD bit pattern that represents the clear-forward state of the transmit line. 1001 The ABCD bit pattern that represents the clear-back state of the receive line. 1101 The ABCD bit pattern that represents the clear-back state of the transmit line. 1101 The ABCD bit pattern that represents the release-guard state of the receive line. 1001 The ABCD bit pattern that represents the release-guard state of the transmit line. 1001 The ABCD bit pattern that represents a blocked receive line. 1101 The ABCD bit pattern that represents a blocked transmit line. 1101 Country mode ITU-T default Example # Restore the default of all the configurations on CAS 1/0/0:5. [H3C-E1 1/0/0] timeslot-set 5 timeslot-list 1-31 signal r2 [H3C-cas 1/0/0:5] default 4-8 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands 4.1.10 delay Syntax delay { call-interval | hold | rising | send-dtmf | dtmf | dtmf-interval | wink-rising | wink-hold | send-wink | start-dial } milliseconds undo delay { call-interval | hold | rising | send-dtmf | dtmf | dtmf-interval | wink-rising | wink-hold | send-wink | start-dial } View Digital E&M voice subscriber-line view Parameter call-interval milliseconds: The call interval for digital E&M voice subscriber-line, in the range of 200 to 2000ms. By default, the value is 200ms. hold milliseconds: The maximum duration the caller will wait for the callee to hang up so as to send the DTMF number in analog E&M subscriber-line delay-start mode. It ranges from 100 to 5000ms, and is defaulted to 400ms. rising milliseconds: The maximum duration the caller will wait from the sending of off-hook signal to the callee's status detected in analog E&M subscriber-line delay-start mode. It ranges from 20 to 2000ms, and is defaulted to 300ms. send-dtmf milliseconds: The delay before the caller sends signal in analog E&M subscriber-line immediate-start mode. It ranges from 50 to 5000ms, and is defaulted to 100ms. dtmf milliseconds: The lasting duration of the DTMF signals in the range of 50 to 500ms. By default, the value is 120ms. dtmf-interval milliseconds: The interval between two DTMF signals in the range of 50 to 500ms. By default, the value is 120ms. wink-rising milliseconds: The maximum duration the caller will wait for the wink signals after sending the seizure signals in analog E&M subscriber-line wink-start mode. It ranges from 100 to 5000ms, and is defaulted to 500ms. wink-hold milliseconds: The maximum lasting duration of the wink signals are received by the caller in analog E&M subscriber-line wink-start mode. It ranges from 100 to 3000ms, and is defaulted to 500ms. send-wink milliseconds: Specifies how long the called party will delay sending wink signal after receiving the seizure signal on E&M interface. It ranges from 100 to 5000ms and defaults to 500ms. start-dial milliseconds: The delay for dialing. It is in the range of 1 to 10 seconds and defaults to 1 second. 4-9 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Description Use the delay command to configure the related time parameters at the digital E&M subscriber-line (E1 controller). Use the undo delay command to restore these parameters to the default value. On the digital E&M subscriber-line, the waiting duration of originating the next call set by the delay call-interval milliseconds command is applicable to the immediate, wink and delay start. When the digital E&M subscriber-line uses the delay start mode, the calling party will use the delay hold milliseconds command to set the longest time waiting for the delay signals . When the digital E&M subscriber-line uses the delay start mode, use the delay rising milliseconds command to set the time waiting for the delay signals of called party after the calling party sends out the off-hook signals, and then detect the device state of the called party. When the digital E&M subscriber-line uses the immediate start mode, use the delay send-dtmf milliseconds command to set the delay time before the calling party sends the called number. The delay dtmf milliseconds command is used to set the duration that the DTMF signals are being sent, and the delay dtmf-interval milliseconds command is used to set the interval for sending the DTMF signals. When the digital E&M subscriber-line uses the wink start mode, the delay wink-rising milliseconds command is used to set the longest time it takes for the calling party to wait for the wink signals after it sends the seizure signals. When the digital E&M subscriber-line uses the wink start mode, the delay wink-hold milliseconds command is used to set the longest duration that the wink signals sent by the called party. When the digital E&M subscriber-line uses the wink start mode, the delay send-wink milliseconds command is used to set the longest time that the called party delays before sending the wink signals. Related command: timeslots-set, timer. Example # Set the longest delay-waiting duration to 3000 milliseconds on voice subscriber line 1/0/1:3. [H3C-voice-line1/0/1:3] delay hold 3000 4-10 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands 4.1.11 display voice em call-statistic Syntax display voice em call-statistic View Any view Parameter None Description Use the display voice em call-statistic command to view the call statistics of the digital E&M subscriber-line. This command applies to E1 voice only. This command is used to display the statistics of the E&M signaling being placed in the state of idle, seizure ready, seizure confirmation, transmitting numbers, receiving numbers, conversation and clear ready. Example # Display the call statistics of the E&M subscriber-line. [H3C] display voice em call-statistic On state of EM_IDLE : On state of EMCALLER_WAIT_OCCUPY : On state of EMCALLER_WAIT_SEND_NUMBER : On state of EMCALLER_SENDING_NUMBER : On state of EMCALLER_RINGING : On state of EMCALLER_TALKING : On state of EMCALLER_CALLER_ONHOOK : On state of EMCALLED_WAIT_SEND_OCCUPY : On state of EMCALLED_WAIT_RECEIVE_NUMBER : On state of EMCALLED_RECEIVING_NUMBER : On state of EMCALLED_RINGING : On state of EMCALLED_TALKING : On state of EMCALLED_CALLED_ONHOOK : On state of EMCALLED_BUSYTONE : On state of STATE_UNKNOWN : 4-11 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Table 4-2 E&M call statistics field description Field Description EM_IDLE Number of messages processed in idle status by E&M module EMCALLER_WAIT_OCCUPY Number of messages processed in waiting for seizure status by E&M module EMCALLER_WAIT_SEND_NUMBER Number of messages processed in the caller end waiting for sending phone number status by E&M module EMCALLER_SENDING_NUMBER Number of messages processed in the caller end sending phone number status by E&M module EMCALLER_RINGING Number of messages processed in the caller end ringback status by E&M module EMCALLER_TALKING Number of messages processed in the caller end talking status by E&M module EMCALLER_CALLER_ONHOOK Number of messages processed in the caller end hooking on status by E&M module EMCALLED_WAIT_SEND_OCCUPY Number of messages processed in the called end waiting for seizure status by E&M module EMCALLED_WAIT_RECEIVE_NUM BER Number of messages processed in the called end waiting for receiving phone number status by E&M module EMCALLED_RECEIVING_NUMBER Number of messages processed in the called end receiving phone number status by E&M module EMCALLED_RINGING Number of messages processed in the called end ringing back status by E&M module EMCALLED_TALKING Number of messages processed in the called end talking status by E&M module EMCALLED_CALLED_ONHOOK Number of messages processed in the called end hooking on status by E&M module EMCALLED_BUSYTONE Number of messages processed in the called end busy status by E&M module STATE_UNKNOWN Number of messages processed unknown status by E&M module 4-12 in Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands 4.1.12 display voice em ccb Syntax display voice em ccb View Any view Parameter None Description Use the display voice em ccb command to view the information of call control block of the E&M subscriber-line. This command applies to E1 voice only. This command is used to display the call status, call-ID, start type, signal type, channel number, etc. Example # Display the information of E&M signaling call control block. [H3C] display voice em ccb EMCCB of channel[ 5]: status : EMCALLER_TALKING em call ID : 1624 start type : EM_START_IMMEDIATE signal type : EM_SIGNAL_DIGITAL channel : 5 rcv call ID : 1639 sig-wait timer : 0 sig-conf timer : 0 msg-wait timer : 0 MCCB of channel[ 37]: status : EMCALLED_TALKING m call ID : 1625 start type : EM_START_IMMEDIATE signal type : EM_SIGNAL_DIGITAL channel : 37 rcv call ID : 1640 sig-wait timer : 0 sig-conf timer : 0 msg-wait timer : 0 4-13 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Table 4-3 E&M call control block field description Field Description status Status of the call em call ID Identifier of E&M module in calling status start type Start mode of E&M connection signal type Signal type of E&M connection channel Channel number of E&M connection rcv call ID Identifier of RCV module in calling status sig-wait timer Timer of waiting for signal sig-conf timer Timer of confirming signal msg-wait timer Timer of waiting for message 4.1.13 display voice r2 call-statistics Syntax display voice r2 call-statistics View Any view Parameter None Description Use the display voice r2 call-statistics command to view the R2 call statistics. This command applies to E1 voice only. Related command: reset voice r2 call-statistics. Example # Display the information of R2 signaling call statistics. [H3C] display voice r2 call-statistics [ E1-Group(0:0) Call Statistics ] +-------------------------------------------------+ [Call sumcount] -> 4 [call success] -> 4 [Call failure] -> 0 4-14 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands [Call-in count] -> 1 [Call-in success] -> 1 [Call-in failure] -> 0 [Call-in answer] -> 0 [Call-in nullnum] -> 0 [Call-out count] -> 3 [Call-out success] -> 3 [Call-out failure] -> 0 [Call-out answer] -> 3 [Call-out busy] -> 0 [Call-out nullnum] -> 0 [Call-out congestion] -> 0 Table 4-4 R2 signaling call statistics field description Field Description Call sumcount Total number of calls: total number of calls in E1 time slot group (sum of incoming and outgoing calls) call success Number of successful calls: total number of successful signaling connection, that is, R2 signaling connection is successfully completed, and the opposite exchange is available for another call connection. Call failure Number of failed calls: total number of failed signaling exchange during call connecting process, such as peer subscriber line busy, null called number, and line failure, etc.) Call-in count Total number of incoming calls Call-in success Total number of successful incoming calls Call-in failure Total number of failed incoming calls Call-in answer Number of sent answers: number of answer signals that the local exchange sends to the originating point when the call is connected successfully and the called party picks up the phone. Call-in nullnum Routing failure times: number of failed incoming calls due to no corresponding route available for the called number to create connection. Call-out count Total number of outgoing calls Call-out success Total number of successful outgoing calls Call-out failure Total number of failed outgoing calls Call-out answer Number of received answers: number of answer signals received from the terminating point when the call is connected successfully. 4-15 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Field Description Call-out busy Subscriber line busy times: number of subscriber line busy signals received from the terminating point during call connecting process. Call-out nullnum Count of null calling numbers: number of null number signals received from the terminating point during call connecting process. Call-out congestion Number of received congestion: number of congestion signals received from the terminating point during call connecting process. 4.1.14 display voice rcv statistic r2 Syntax display voice rcv statistic r2 View Any view Parameter None Description Use the display voice rcv statistic r2 command to view the information of call statistics related to the R2 signaling in the RCV module. This command applies to E1 voice only. This command is used to display the interactive messages between the RCV module and the R2 signaling module, including the number of sending the messages of the connection request to acknowledge success and fail, the number of sending the messages of activation acknowledgement of success and failure, the number of sending the messages of on-hook and off-hook, the number of receiving the messages of connection request, the number of receiving activation messages, and the number of receiving such messages as release, ringing, and unknown. Example # Display the information of R2 signaling call statistics in RCV module . [H3C] display voice rcv statistic r2 Statistic between RCV and R2 : { Send_R2_ConnectReqAck_SUCCESS : 0 Send_R2_ConnectReqAck_FAIL : 0 4-16 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Send_R2_ActiveAck_SUCCESS : 0 Send_R2_ActiveAck_FAIL : 0 Send_R2_Onhook : 0 Send_R2_Offhook : 0 Send_R2_IPAlerting : 0 Recv_R2_ConnectReq : 0 Recv_R2_Active_TD_IN : 0 Recv_R2_Active_TD_OUT : 0 Recv_R2_Active_ELSE : 0 Recv_R2_Release : 0 Recv_R2_Alert_AP_ALERTING : 0 Recv_R2_Alert_ELSE : 0 Recv_R2_Unknow : 0 } Table 4-5 Field description of R2 signal call statistics in RCV module Field Description Send_R2_ConnectReqAck_SUC CESS Number of sending connection request success-acknowledge message to R2 module Send_R2_ConnectReqAck_FAIL Number of sending connection request failure-acknowledge message to R2 module Send_R2_ActiveAck_SUCCESS Number of sending activation success-acknowledge message to R2 module Send_R2_ActiveAck_FAIL Number of sending activation failure-acknowledge message to R2 module Send_R2_Onhook Number of sending onhook message to R2 module Send_R2_Offhook Number of sending offhook message to R2 module Send_R2_IPAlerting Number of sending IP side alerting message to R2 module Recv_R2_ConnectReq Number of receiving R2 connection request message Recv_R2_Active_TD_IN Number of receiving R2 called end hooking off message Recv_R2_Active_TD_OUT Number of receiving R2 caller end hooking off message Recv_R2_Active_ELSE Number of receiving R2 other hooking off message Recv_R2_Release Number of receiving R2 release request message Recv_R2_Alert_AP_ALERTING Number of receiving R2 alerting message 4-17 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Field Description Recv_R2_Alert_ELSE Number of message receiving R2 other alerting Recv_R2_Unknow Number of receiving R2 unknown message 4.1.15 display voice subscriber-line Syntax display voice subscriber-line e1-number : { ts-set-number | 15 } display voice subscriber-line t1-number : 23 View Any view Parameter e1-number, t1-number: Indicates the number of subscriber line generated in creating the Timeslot set or the ISDN PRI set. ts-set-number: Indicates the number of the Timeslot set created successfully. 15: Indicates the subscriber line is generated in creating the ISDN PRI set on E1 subscriber line. 23: Indicates the subscriber line is generated in creating the ISDN PRI set on T1 subscriber line. Description Use the display voice subscriber-line command to view the subscriber line configuration. This command applies to E1 and T1 voice. The command display voice subscriber-line e1-number:ts-set-number is mainly used to display the Timeslot set corresponding to the E1 subscriber line, whether to adopt the private-line auto-ring connection and connection number, the subscriber line description, whether to start the echo cancellation function and echo cancellation sampling time length, and whether to start the comfort noise function. The command display voice subscriber-line e1-number:15 is mainly used to display the configuration information of the subscriber line corresponding to the ISDN PRI set, the command display voice subscriber-line t1-number:23 is mainly used to display the configuration information of the subscriber line corresponding to the ISDN PRI set, such as whether to adopt the private-line auto-ring connection, subscriber line description, whether to start the echo cancellation function and set echo cancellation sampling time length, input gain and output attenuation, whether to adopt the nonlinear 4-18 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands processing in echo cancellation, the time of waiting for the initial number and the dial-up time interval between numbers, etc. Example # Display the configuration about voice subscriber-line 1/0/0:0. [H3C] display voice subscriber-line 1/0/0:0 The voice line was ds0 this subscriber line was not set connection The subscriber line's description: echo cancellation enable echo cancellation coverage 16 comfort noise enable PCM companding type :A-law [H3C] display voice subscriber-line 1:15 The voice line was pri this subscriber line was not set connection The subscriber line's description: echo cancellation enable echo cancellation coverage 16 music threshold is -38 receive gain 0 transmit gain 0 first-dial timer 10 dial-interval Timer 10 PCM companding type :A-law Table 4-6 Field description on voice subscriber-line configuration Field Description The voice line The signaling type of the subscriber line this subscriber line The connection method of the subscriber line The subscriber line's description The description of the subscriber line echo cancellation The echo cancellation configuration of the subscriber line echo cancellation coverage The echo cancellation coverage of the subscriber line comfort noise The comfort noise of the subscriber line The voice line was The signaling type of the subscriber line this subscriber line The connection method of the subscriber line receive gain The receive gain of the subscriber line transmit gain The transmit gain of the subscriber line 4-19 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Field Description first-dial timer The timeout value after the subscriber dials the first digit dial-interval Timer The timeout value between dialing the first digit and dialing the second digit 4.1.16 display voice voip Syntax display voice voip { down-queue e1t1vi-no | phy-statistic e1t1vi-no | up-queue e1t1vi-no | } View Any view Parameter e1t1vi-no: Indicates the card number of the E1/T1 voice card. Description This command applies to E1 and T1 voice. The command display voice voip downqueue e1t1vi-bno displays the contents of the down interrupt queue between the E1/T1 voice card and the router main card. The command display voice voip up-queue e1t1vi-no displays the contents of the up interrupt queue between the E1/T1 voice card and the router main card. The command display voice voip phy-statistic e1t1vi-bno displays the statistics of the physical layer. Example # Display the contents of the down interrupt queue between the E1/T1 voice card and the router main card. [H3C] display voice voip down-queue 5 V = 0,I = 0,P = 0,C = 0,E = E1/T1VI_NULL_EVENT, B = 0 V = 0,I = 1,P = 0,C = 0,E = E1/T1VI_NULL_EVENT, B = 0 V = 0,I = 2,P = 0,C = 0,E = E1/T1VI_NULL_EVENT, B = 0 …… V = 0,I = 255,P = 0,C = 0,E = E1/T1VI_NULL_EVENT, B = 0 E1VI board 0 down interrupt queue is empty : 4-20 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Table 4-7 Field description of the down interrupt queue between the E1/T1 voice card and the router main card. Field Description V Value flag of interrupt I Sequence number of interrupt down-queue P Port number of E1VI board C Channel number of E1VI board E Type of event B Flag of queue ending E1VI board 0 down interrupt queue is empty Interrupt down-queue of E1VI board is empty 4.1.17 dl-bits Syntax dl-bits { answer | blocking | clear-back | clear-forward | idle | seize | seizure-ack | release-guard } { receive | transmit } ABCD undo dl-bits { answer | blocking | clear-back | clear-forward | idle | seize | seizure-ack | release-guard } { receive | transmit } View R2 CAS view Parameter answer: Answer signal of R2 line signaling. blocking: Blocking signal of R2 line signaling. clear-back: Clear-back signal of R2 line signaling. clear-forward: Clear-forward signal of R2 line signaling. idle: Idle signal of R2 line signaling. seize: Seizure signal of R2 line signaling. seizure-ack: Seizure acknowledgement signal of R2 line signaling. release-guard: Release guard signal of R2 line signaling. receive: Signal of receiving R2 line signaling transmit: Signal of sending R2 line signaling ABCD: Value of signal bits of receiving/sending R2 line signaling, ranging from 0000 to 1111. 4-21 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Table 4-8 Default values of R2 line signaling Signal Default rx-bits ABCD Default tx-bits ABCD Answer 0101 0101 Blocking 1101 1101 Clear-back 1101 1101 Clear-forward 1001 1001 Idle 1001 1001 Seize 0001 0001 Seizure-ack 1101 1101 Release-guard 1001 1001 Description Use the dl-bits command to configure the bit value of all the signals of R2 signaling. Use the undo dl-bits command to restore the values to defaults. This command applies to E1 voice only. You can use the dl-bits command to configure the ABCD bits value of R2 signaling for different coding schemes in different countries. For countries other than China, keep the value of the CD bits to be “01”. When you modify a bit value of R2 signaling, you need to simultaneously modify the values of other bits so that the whole system can work normally in the new signaling state machine. Related command: seizure-ack {enable/disable} and answer {enable/disable}. Example # Configure ABCD bits value of the idle signal of receiving R2 signaling to 1101, and that of the idle signal of transmitting R2 signaling to 1011. [H3C-cas1/0/4:0] dl-bits idle receive 1101 [H3C-cas0:0] dl-bits idle transmit 1011 4.1.18 dtmf Syntax dtmf { enable | disable } View R2 CAS view 4-22 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Parameter enable: Enables R2 signaling to be received and sent in DTMF mode. disable: Disables R2 signaling to be received and sent in DTMF mode. Description Use the dtmf command to configure the way of sending and receiving R2 signaling. By default, MFC mode, not DTMF mode, is used to collect call number information. This command applies to E1 voice only. The dtmf command is used to configure whether to use MFC or DTMF for sending and receiving R2 signaling. DTMF mode is used when you configure dtmf enable. MFC mode is used when you configure dtmf disable. Related command: timer dtmf. Example # Configure DTMF mode to receive and send R2 signaling. [H3C-cas1/0/0:0] dtmf enable 4.1.19 effect-time Syntax effect-time time undo effect-time View R2 CAS view Parameter time: The lower threshold of the debounce time of line signaling, in the range 10 to 40 milliseconds. It defaults to 10 milliseconds. Description Use the effect-time command to configure the debounce time of line signaling. Use the undo effect-time command to restore the default. This command applies to E1 voice only. The change of the line is regarded valid only when the duration of line signaling exceeds this threshold. Related command: timeslots-set, pri-set. 4-23 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example # Set the debounce time of line signaling to 20 milliseconds for timeslot set 3 on the E1 subscriber-line 1/0/0. [H3C-cas1/0/0:3] effect-time 20 4.1.20 final-callednum Syntax final-callednum { enable | disable } View R2 CAS view Parameter enable: Enables the called number terminate signal to be sent back. disable: Disables the called number terminate signal to be sent back. Description Use the final-callednum command to enable or disable the terminate signal to be sent to the terminating point after the called number is sent. By default, the called number terminate signal is disabled. This command applies to E1 voice only. In some countries, the R2 interregister signaling can be used to send the called number terminate signal after the called number is sent out, indicating that the called number transmission is completed. In this case, you can use this command to adjust the signaling exchange approach. When the terminating point receives the terminate signal, it stops requesting for the called number. Related command: register-value digital-end. Example # Enable the called number terminate signal. [H3C-cas1/0/0:0] final-callednum enable 4.1.21 force-metering Syntax force-metering { enable | disable } View R2 CAS view 4-24 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Parameter enable: Enables the metering signal of R2 signaling. disable: Disables the metering signal of R2 signaling. Description Use the force-metering command to enable or disable the metering signal of R2 signaling. By default, the metering signal of R2 signaling is disabled. This command applies to E1 voice only. If the opposite exchange supports the metering signal, H3C router, as the terminating point, sends a forced-release signal instead of a clear-back signal when it terminates a call, so as to indicate that the called party has release the line and the call is terminated. And so metering signal collision can be avoided. Example # Enable the metering signal of R2 signaling. [H3C-cas1/0/0:0] force-metering enable 4.1.22 group-b Syntax group-b { enable | disable } View R2 CAS view Parameter enable: Enables signal exchange at Group-B stage of R2 signaling. disable: Disables signal exchange at Group-B stage of R2 signaling, that is, backward Group-A signal A6 is used to complete register exchange directly. Description Use the group-b command to enable or disable Group-B stage signal to complete register exchange. By default, Group-B stage signal is used to complete register exchange, that is, the command is in enable state. This command applies to E1 voice only. 4-25 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands In some countries, R2 register exchange does not support Group-B stage signal exchange, or cannot correctly interpret Group-B signal value. Then you can use the group-b command to enable or disable Group-B signal exchange. Related command: register-value req-switch-groupb. Example # Enable Group-B signal to complete register exchange. [H3C-cas1/0/0:0] group-b enable 4.1.23 line Syntax line slot -number : { ts-set-number | 15 } line slot -number : 23 undo line View POTS voice entity view Parameter slot –number: Indicates the serial number of E1/T1 subscriber-line to which this subscriber line belongs. ts-set-number: Indicates the serial number of Timeslot set established successfully. 15: Indicates adopting the E1 voice ISDN PRI interface mode. 23: Indicates adopting the T1 voice ISDN PRI interface mode. Description Use the line command to configure the corresponding relationship between the POTS voice entity and the logic subscriber line. Use the undo line command to cancel the corresponding relationship between the POTS peer and the logical subscriber line. This command applies to E1 and T1 voice. This command can be used in POTS voice entity view only and takes effect on E1/T1 voice subscriber-line only. After configuring the destination mode of voice entity by using the command match-template, it is required to use the command line to configure the corresponding relationship between the POTS voice entity and logic line, that is, to specify that via which line the routing should be performed toward this destination. 4-26 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Use the command line e1-number:ts-set-number to select the subscriber line corresponding to Timeslot set in this E1 subscriber-line as the routing output. Use the command line e1-number:15 corresponding to the ISDN PRI set in this E1 subscriber-line as the routing output. Use the command line t1-number:23 corresponding to the ISDN PRI set in this T1 subscriber-line as the routing end. Related command: timeslots-set, entity, pri-set. Example # Configure the corresponding relationship between POTS voice entity 3 and No.1 Timeslot set in E1 subscriber-line. [H3C-voice-dial-entity3] line 1/0/1:1 4.1.24 loopback Syntax loopback { local | remote | payload } undo loopback View CE1/PRI interface view Parameter local: Places the local physical layer in a local loopback. remote: Places the local physical layer in a remote loopback. payload: Places the local physical layer in a payload loopback. Description Use the loopback command to set the loopback mode on the interface. Use the undo loopback command to restore the default. Example # Place the local physical layer of interface E1 1/0 in a local loopback. [H3C-E1 1/0/0] loopback local 4.1.25 mfc (R2 CAS) Syntax mfc { block | open | query } timeslots timeslots-list 4-27 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands View R2 CAS view Parameter block: Indicates blocking the MFC channel of the specified timeslot. open: Indicates opening the MFC channel of the specified timeslot. query: Indicates querying the MFC channel of the specified timeslot. timeslots-list: Specifies a timeslot range. The expression includes the single digit, two digits separated by “,”, a pair of digits separated by “-”, or the combination form (e.g., 1-14, 15, 17-31). The value range of digits is the integer from 1 to 31. Description Use the mfc command to maintain of MFC channel of the specified timeslot. This command applies to E1 voice only. To block the MFC channel means that this channel will no longer load the R2 interregister signaling information, that is, this channel is set manually as unavailable. To open the MFC channel is the processes inverse to the blocking operation, which can re-set the channel as available and enable it to load R2 interregister signaling. To query the MFC channel will display the busy/idle, opened/blocked status of channel in a real time way. Related command: cas, ts. Example # Block the timeslots 1-15 in No.5 Timeslot set, and query the channel status of the timeslots 1-31. [H3C-cas1/0/0:5] mfc block timeslots 1-15 [H3C-cas1/0/0:5] mfc query timeslots 1-31 4.1.26 mode Syntax mode zone-name [ default-standard ] undo mode View R2 CAS view 4-28 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Parameter zone-name: Name of the country or region. By default, it is set to the mode of ITU-T. It can be: z argentina: Uses Argentinean R2 signaling standard. z australia: Uses Australian R2 signaling standard. z china: Uses Chinese R2 signaling standard. z bengal: Uses Bengalee R2 signaling standard. z brazil: Uses Brazilian R2 signaling standard. z custom: Uses the R2 signaling mode defined by customer. z hongkong: Uses Hongkong R2 signaling standard. z india: Uses Indian R2 signaling standard. z indonesia: Uses Indonesian R2 signaling standard. z itu-t: Uses ITU-T R2 signaling standard. z korea: Uses Korean R2 signaling standard. z malaysia: Uses Malaysian R2 signaling standard. z mexico: Uses Mexican R2 signaling standard. z newzealand: Uses New Zealand R2 signaling standard. z singapore: Uses Singaporean R2 signaling standard. z thailand: Uses Thai R2 signaling standard. default-standard: Initializes the related parameters of R2 signaling according to the mode in the current country, that is, initializes the values of the sense and force-metering commands. Description Use the mode command to configure the R2 signaling mode in a country or region. Use the undo mode command to restore the default, as you would with the mode itu-t default-standard command. The default R2 signaling mode is ITU-T mode. This command applies to E1 voice only. The implementation and parameters of R2 signaling vary in different countries. Therefore, the mode needs to be adjusted to enable H3C router to exchange R2 signaling with the switching devices of different countries or regions. According to the configuration, the system automatically selects the appropriate subscriber-line state, service type, metering signal, and the signal values of C and D bits, etc. At present, it supports the modes in Brazil, Mexico, Argentina, India, New Zealand, Thailand, Bengal, South Korea, Hongkong, Indonesia, as well as the countries and regions that comply with ITU-T standards. If the default-standard parameter is configured, then the system will initialize the subscriber line status, service type, metering signal and C signaling bit and D signaling 4-29 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands bit and other parameters according to the national standards of the countries or regions. In the custom mode, you can configure the specific signaling exchange process and signal values of R2 signaling, so as to adjust the R2 signaling in your country in a more flexible way. You can execute this command only when the register-enable off command is configured. Related command: register-value, force-metering, effect-time. Example # Adopt the default standard of Hongkong R2 signaling mode. [H3C-cas1/0/0:0] mode hongkong default-standard 4.1.27 open-trunk Syntax open-trunk { caller [ monitor interval ] | called } undo open-trunk [caller monitor ] View E&M voice subscriber line view Parameter caller: Indicates the voice gateway can enable E&M non-signaling mode but cannot enable monitoring function when it works as the calling end. monitor interval: Indicates the voice gateway can enable both E&M non-signaling mode and the monitoring function when it works as the calling end. The monitoring time is decided by the argument interval. called: Indicates the voice gateway can enable E&M non-signaling mode when it works as the called end. Description Use the open-trunk command to enable E&M non-signaling mode. Use the undo open-trunk command to delete E&M non-signaling mode. By default, E&M non-signaling mode is disabled. 4-30 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Note: After using the open-trunk caller monitor interval command to enable E&M non-signaling mode and enable the monitoring function, you should use the undo open-trunk monitor command to delete this monitoring function (Note that the execution of this command will just disable monitoring function but not E&M non-signaling mode.); while executing the undo open-trunk command will not only disable E&M non-signaling mode but also the monitoring function. Example # Configure to enable E&M non-signaling mode and enable the monitoring function at the same time. The monitoring time is 120 seconds. [H3C-voice-line3/0/0] open-trunk caller monitor 120 4.1.28 pcm Syntax pcm { a-law | µ-law } undo pcm View Voice subscriber-line view Parameter a-law: Companding A-law, used in most part of the world other than North America and Japan, such as China, Europe, Africa, and South America. µ-law: Companding µ-law, used in North America and Japan. Description Use the pcm command to configure a companding law used for quantizing signals. Use the undo pcm command to restore the default. By default, A-law applies. Companding laws are usually adopted to quantize signals unevenly for the purpose of reducing noise and improving signal to noise ratio. Underpinning this approach is the statistics about voice signals, which indicate that lower power signals are more likely present than high power signals. According to CCITT, when two countries use different companding schemes communicate, the side using µ-law is responsible for converting signals to A-law. 4-31 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example # Adopt µ-law companding for signal quantization. [H3C-voice-line3/0/0:0] pcm u-law 4.1.29 pri-set Syntax pri-set [ timeslot-list range ] undo pri-set View CE1/PRI interface view, CT1/PRI interface view Parameter range: Number of bundled timeslots. It ranges from 1 to 31 in CE1/PRI interface view and from 1 to 24 in CT1/PRI interface view. When specifying the timeslots to be bundled, you can specify one timeslot by specifying its number, a timeslot range by specifying a range in the form number1-number2, or several discrete timeslots in the form number1 number2-number3. Description Use the pri-set command to bundle timeslots on the CT1/PRI or CE1/PRI interface into a pri-set. Use the undo pri-set command to remove the timeslot bundle. By default, no timeslots are bundled into pri-sets. When creating a pri-set on a CE1/PRI interface, note the following: z Timeslot 16 is D channel for transmitting signaling; it cannot form a bundle that includes itself only. The attempt to bundle only timeslot 16 will fail. z In a pri-set formed by timeslot bundling on a CE1/PRI interface, timeslot 0 is used for frame synchronization control (FSC), timeslot 16 as a D channel for signaling transmission, and other timeslots as B channels for data transmission. You may bundle the timeslots except for timeslot 0 into a pri-set (as the D channel, timeslot 16 is automatically bundled). The logic features of this pri-set will be the same like those of an ISDN PRI interface. If no timeslot is specified, all timeslots except for timeslot 0 are bundled into an interface similar to an ISDN PRI interface in the form of 30B+D. z The system automatically creates a serial interface after timeslot bundling on the interface. This serial interface has the same logic features of ISDN PRI interface. The serial interface is numbered in the form of serial number:15, where number is maximum serial interface number plus 1. 4-32 Command Manual – Voice Comware V3 z Chapter 4 E1/T1 Voice Configuration Commands Only one timeslot bundling mode is supported on one CE1/PRI interface at a time. In other words, you cannot use this command and the channel-set command together. When creating a pri-set on a CT1/PRI interface, note the following: z Timeslot 24 is D channel for transmitting signaling; it cannot form a bundle that includes itself only. The attempts to bundle only timeslot 24 will fail. z In a pri-set formed by timeslot bundling on a CT1/PRI interface, timeslot 24 is used as D channel for signaling transmission, and other timeslots as B channels for data transmission. You may randomly bundle these timeslots into a pri-set (as the D channel, timeslot 24 is automatically bundled). The logic features of this pri-set will be the same like those of an ISDN PRI interface. If no timeslot is specified, all the timeslots are bundled into an interface similar to an ISDN PRI interface in the form of 23B+D. z The system automatically creates a serial interface after timeslot bundling on the interface. This serial interface has the same logic features of ISDN PRI interface. The serial interface is numbered in the form of serial number:23, where number is maximum serial interface number plus 1. z Only one timeslot bundling mode is supported on a CT1/PRI interface at a time. In other words, you cannot use this command and the channel-set command together. Example # On the CE1/PRI interface bundle timeslots 1, 2, and 8 through 12 into a pri-set. [H3C-E1 3/0/0] pri-set timeslot-list 1,2,8-12 4.1.30 re-answer Syntax re-answer { enable | disable } View R2 CAS view Parameter enable: Enables the originating point to support re-answer signal process. disable: Disables the originating point to support re-answer signal process. Description Use the re-answer command to enable or disable the originating point to support re-answer signal process. By default, the originating point does not support re-answer signal process. 4-33 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands This command applies to E1 voice only. In some countries, re-answer process is needed in R2 signaling. When the terminating point sends a clear-back signal, the originating point does not release the line right away, but maintains the call state instead. If it receives the re-answer signal from the terminating point in the specified time, it continues the call; otherwise, it disconnects the call after timeout. Related command: answer, timer dl re-answer. Example # Enable the originating point to process re-answer signals. [H3C-cas1/0/0:0] re-answer enable 4.1.31 register-number Syntax register-number undo register-number View Voice entity view Parameter None Description Use the register-number command to enable the gateway (the VoIP-enabled router) to register the number of the voice entity with the H.323 gatekeeper or SIP server. Use the undo register-number command to disable the gateway to register the number of the voice entity when registering with the H.323 gatekeeper or SIP server. By default, the number of the voice entity is registered. You need to use the undo register-number command to disable the gateway to register the numbers of some voice entities with a gatekeeper or SIP server in the situations where: z POTS entities with the same number exist on multiple gateways on the same network. But for their gatekeeper or SIP server, the number of a POTS entity must be unique. Therefore, those POTS entities cannot register with the gatekeeper or SIP server at the same time. z The user needs to register the numbers of some ports with the gatekeeper or SIP server but not the numbers of others for special purposes. 4-34 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands This command is only applicable to the configurations of POTS voice entities. Caution: After configuring this command, use the ras-on command in GK-client view or the register-enable command in SIP view to register the number with the gatekeeper or SIP server to make the configuration effective. Related command: match-template. Example # Disable the gateway to register the number of voice entity 1. [H3C-voice-dial-entity1] undo register-number 4.1.32 register-value Syntax register-value { billingcategory | callcreate-in-groupa | callingcategory | congestion | demand-refused | digit-end | nullnum | req-billingcategory | reqcallednum-and-switchgroupa | req-callingcategory | req- currentcallednum-in-groupc | req-currentdigit | req- firstcallednum-in-groupc | req-firstcallingnum | req-firstdigit | req-nextcallednum | req-nextcallingnum | req-lastfirstdigit | req-lastseconddigit | req-lastthirddigit | req- nextcallednum | req-nextcallingnum | req-switch-groupb | subscriber-abnormal |subscriber-busy | subscriber-charge |subscriber-idle } value undo { billingcategory | callcreate-in-groupa | callingcategory | congestion | demand-refused | digit-end | callednum-and-switchgroupa nullnum | | req-billingcategory req-callingcategory | | reqreq- currentcallednum-in-groupc | req-currentdigit | req- firstcallednum-in-groupc | req-firstcallingnum | req-firstdigit | req-nextcallednum | req-nextcallingnum | req-lastfirstdigit | req-lastseconddigit | req-lastthirddigit | req- nextcallednum | req- nextcallingnum | req- specialsignal | req-switch-groupb subscriber-abnormal |subscriber-busy | subscriber-charge |subscriber-idle } View R2 CAS view 4-35 | Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Parameter billingcategory value: Specifies the billing category value, in the range 1 to 16. It defaults to 1. callcreate-in-groupa value: Specifies the directly created calling signal value, in the range 1 to 16. It defaults to 6. callingcategory value: Specifies calling category value, in the range 1 to 16. It defaults to 1. congestion value: Specifies congestion value, in the range 1 to 16. It defaults to 4. demand-refused value: Specifies demand-refused value, in the range 1 to 16. It defaults to 12. digit-end value: Specifies digit-end value, in the range 1 to 16. It defaults to 15. nullnum value: Specifies null number value, which ranges from 1 to 16. It defaults to 5. req-billingcategory value: Specifies request billing category value, in the range 1 to 16. It defaults to 3. req- callednum-and-switchgroupa value: Specifies request next called number and switch Group-A value, in the range 1 to 16. It defaults to 1. req-callingcategory value: Specifies request calling category value, in the range 1 to 16. It defaults to 3. req- currentcallednum-in-groupc value: Specifies request current called number value in Group C, in the range 1 to 16. It defaults to 16. req-currentdigit value: Specifies request current digit value, in the range 1 to 16. It defaults to 16. req- firstcallednum-in-groupc value: Specifies request first called number value in Group C, in the range 1 to 16. It defaults to 16. req-firstcallingnum value: Specifies request first calling number value, in the range 1 to 16. It defaults to 5. req-firstdigit value: Specifies request first digit value, in the range 1 to 16. It defaults to 16. req-nextcallednum value: Specifies request next called number value, in the range 1 to 16. It defaults to 1. req-nextcallingnum value: Specifies request next calling number value, in the range 1 to 16. It defaults to 5. req-lastfirstdigit value: Specifies request last first digit value, in the range 1 to 16. It defaults to 2. req-lastseconddigit value: Specifies request last second digit value, in the range 1 to 16. It defaults to 7. 4-36 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands req-lastthirddigit value: Specifies request last third digit value, in the range 1 to 16. It defaults to 8. req- nextcallednum value: Specifies request next called number value, in the range 1 to 16. It defaults to 1. req- nextcallingnum value: Specifies request next calling number value, in the range 1 to 16. It defaults to 5. req-switch-groupb value: Specifies request switch Group-B value, in the range 1 to 16. It defaults to 3. subscriber-abnormal value: Specifies subscriber line abnormal value, in the range 1 to 16. It defaults to 16. subscriber-busy value: Specifies subscriber line busy value, in the range 1 to 16. It defaults to 3. subscriber-charge value: Specifies subscriber line idle and charge value, in the range 1 to 16. It defaults to 1. subscriber-idle value: Specifies subscriber line idle value, in the range 1 to 16. It defaults to 6. Description Use the register-value command to configure the value of all register signals of R2 signaling. Use the undo register-value command to restore the default value. Signal value 16 means the corresponding signal function does not exist. For example, if the request last first digit function does not exist in some countries, the value of req-lastfirstdigit is 16. This command applies to E1 voice only. By configuring the register-value command, you can send the specified request signal and require the opposite exchange to send back the corresponding answer signal. For example, using the register-value callingcategory command, you can enable the terminating point to send the specified signal to require the originating point to send back the calling category. The register-value billingcategory command is used to configure KA signal of R2 signaling. That is, the originating point sends a calling category signal to the originating toll office or originating international exchange, which provides two kinds of information: billing category of this connection (paid at the specified time or at once, or toll free) and subscriber level (with or without priority). The register-value callingcategory command is used to configure KD signal of R2 signaling, i.e., calling category. It functions to identify whether break-in and forcedrelease can be implemented by or on the calling party. The register-value subscriber-idle command is used to configure KB signal of R2 signaling. It indicates subscriber status (such as idle), and acknowledges and controls 4-37 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands connection. You should make sure that KB values of the both ends are the same. Otherwise, call connection might not be established even if the called party is idle. If H3C routers are used at both ends, you should make sure that KB values of the both ends are the same. If PBX is used at one end, and a router is used at the other end, you should adjust the KB value of the router to keep it consistent with that of the PBX. Note: In some countries, the register signal encoding scheme does not necessarily support all the register signals described above. For example, there is no billingcategory but callingcategory in ITU-T recommendation. Therefore, it is recommended that you use the default values and do not configure unless there are special requirements. Related command: group-b {enable/disable}. Example # Request the originating point to send calling category by configuring a backward signal (signal value 7). [H3C-cas1/0/0:0] register-value req-callingcategory 7 4.1.33 renew Syntax renew ABCD undo renew View R2 CAS view Parameter ABCD: Indicates the default of each signal bit in transmission, with the value being 0 or 1. It defaults to 1111. Description Use the renew command to configure the signal values of C bit and D bit. Use the undo renew command to restore the default value. This command applies to E1 voice only. In the R2 signaling A-bit and B-bit are used to transmit the valid information, and the actual transmission signal has nothing to do with the setting value. C-bit and D-bit do not transmit the valid information, and generally the set signal value is adopted as the 4-38 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands transmission signals. Therefore, for the R2 signaling, this command only makes sense to C-bit and D-bit. Using this command, you can adjust the values of C and D bits according to the line signaling encoding standards in different countries. For example, in China, the values of C and D bits of R2 signaling are fixed to 1 and 1 respectively. However, in most other countries, the values of C and D bits are 0 and 1 respectively. Related command: cas, reverse. Note: This command takes effect only after the mode zone-name {custom| standard} command is configured , and is invalid after the command mode custom is configured, namely the values of C and D will not be changed. With mode custom configured, all signal values of R2 signaling are set manually. With mode zone-name custom configured, some signals values are set as required, while most signal values comply with the standard of the zone-name country. Example # Configure the signal values of both the C-bit and D-bit of R2 line signaling as 1. [H3C-cas1/0/0:5] renew 0011 4.1.34 reset voice em Syntax reset voice em View User view Parameter None Description Use the reset voice em command to reset the call statistics on the digital E&M interface. This command applies to E1 voice lines only. Related command: display voice em call-statistics. 4-39 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example # Reset the call statistics on the E&M interface. reset voice em 4.1.35 reset voice r2 Syntax reset voice r2 View User view Parameter None Description Use the reset voice r2 command to reset the call statistics of R2 signaling. This command applies to E1 voice only. Related command: display voice r2 call-statistics. Example # Reset the call statistics of R2 signaling. reset voice r2 4.1.36 reverse Syntax reverse ABCD undo reverse View R2 CAS view Parameter ABCD: Indicates whether to perform the inversion of each signal bit, with the value of each bit being 0 or 1. The default ABCD is 0000, that is, the function of inversion is disabled. Description Use the reverse command to configure the inversion mode of line signals. 4-40 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Use the undo reverse command to restore the default value. This command applies to E1 voice only. This command can be used to perform the inversion change to A, B, C, and D bits prior to sending and after receiving the line signal, that is, 0 is changed to 1, and 1 to 0. If the value of one bit is 1, it indicates this bit is needed to invert. Related command: cas, renew. Example # reverse the B-bit and D-bit of the R2 line signaling. [H3C-cas1/0/0:0] reverse 0101 4.1.37 seizure-ack Syntax seizure-ack { enable | disable } View R2 CAS view Parameter enable: Enables the terminating point to send seizure acknowledgement signal. disable: Disables the terminating point to send seizure acknowledgement signal. Description Use the seizure-ack command to enable or disable the terminating point to send seizure acknowledgement signal. By default, the terminating point sends seizure acknowledgement signal. This command applies to E1 voice only. Normally, the terminating point sends seizure acknowledgement signal when it receives a seizure signal from the originating point. However, in some countries R2 signaling encoding scheme allows the terminating point not to send seizure acknowledgement signal. In this case, you can use the seizure-ack command to adjust the signaling encoding scheme. If the originating point does not require acknowledgement signal, the terminating point has no need to send back the signal when it receives a seizure signal. Related command: timer dl seizure. Example # Disable the terminating point to send seizure acknowledgement signal. [H3C-cas1/0/0:0] seizure-ack disable 4-41 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands 4.1.38 select-mode Syntax select-mode [ max | maxpoll | min | minpoll ] undo select-mode View R2 CAS view Parameter max: Indicates the maximum selection. maxpoll: Indicates the maximum polling selection. min: Indicates the minimum selection. minpoll: Indicates the minimum polling selection. Description Use the select-mode command to configure the E1 trunk selection mode. Use the undo select-mode command to restore the default. By default, the trunk selection mode is min. This command applies to E1 voice only. The proper selection policy can not only enables each timeslot in E1 trunk to have the balanced opportunity to be used but also helps to enhance the speed of selecting idle timeslot, so as to improve the telephone connection speed. The parameter max indicates the maximum selection, selecting the timeslot of the maximum serial number from the currently available timeslots. The parameter maxpoll indicates the maximum polling selection. When used for the first time, select the timeslot of the maximum serial number from the currently available timeslots, and next time select in descending order the available timeslot whose serial number is less than it. For example, No.31 and No.29 timeslots are unavailable in 32 timeslots, so select firstly No.30 timeslot, and select No.28 timeslot secondly. The parameter min indicates the minimum selection, selecting the timeslot of the minimum serial number from the currently available timeslots. The parameter minpoll indicates the minimum polling selection. When used for the first time, select the timeslot of the minimum serial number from the currently available timeslots , and next time select in descending order the available timeslot whose serial number is bigger than it. For example, No.1 and No.3 timeslots are unavailable in 32 timeslots, so select firstly No.2 timeslot, and select No.4 timeslot secondly. Using the command select-mode without any parameter can restore the default trunk timeslot selection mode. 4-42 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Related command: cas and trunk-direction. Example # Configure the trunk selection mode as max for the No.5 Timeslot set in No.0 E1 subscriber-line. [H3C-cas1/0/0:5] select-mode max 4.1.39 send-dialtone Syntax send-dialtone undo send-dialtone View Voice subscriber line view Parameter None Description Use the send-dialtone command to enable the voice subscriber line to play the dial tone to the PSTN side immediately if a SETUP message indicating a vacant called number is received. Use the undo send-dialtone command to disable the voice subscriber line from playing the dial tone to the PSTN side. By default, no dial tone is played to the PSTN side. Example Enable the voice subscriber line to play the dial tone to the PSTN side if a SETUP message indicating a vacant called number is received from the PBX of the PSTN. [H3C-voice-line1/0:15]send-dialtone 4.1.40 sendring Syntax sendring { ringback | ringbusy } { enable | disable } View R2 CAS view 4-43 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Parameter ringback { enable | disable }: Enables/disables the sending of ring-back signal. ringbusy { enable | disable }: Enables/disables the sending of ring-busy signal. Description Use the sendring command to enable or disable the terminating point to send ring-back tone or busy tone signal to the calling party. By default, the ring-back tone and busy tone signals are sent. This command applies to E1 voice only. In some regions a PBX, as the originating point, might not send ring-back tone to the calling party during call connecting process. To avoid call connection failure due to the calling party not able to hear the corresponding tone, you can manually configure the sendring command. If H3C router works as the terminating point, it sends the corresponding tone to the originating point according to call connection situations. Related command: timer ringring. Example # When time slot 5 on the E1 port 0 works as the terminating point, enable it to send ring-back tone to the originating point. [H3C-cas1/0/0:5] sendring ring-back enable 4.1.41 signal-value Syntax signal-value { received idle | received seize | transmit-bits idle | transmit seize } ABCD undo signal-value { received idle | received seize | transmit-bits idle | transmit seize } View Digital E&M voice subscriber-line view Parameter received idle: The digital E&M subscriber-line receives idle signaling. received seize: The digital E&M subscriber-line receives the seizure signaling. transmit idle: That the digital E&M subscriber-line transmits the idle signaling. transmit seize: The digital E&M subscriber-line transmits the seizure signaling. 4-44 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands ABCD: The default value of each signaling bit in a transmission. It is valued to be either 0 or 1. By default, the ABCD bits value of the received and transmitted idle signaling and seizure signaling on the digital E&M subscriber-line are 1101. Description Use the signal-value command to configure the digital E&M subscriber-line to receive and transmit the ABCD bits of idle signaling and seizure signaling. Use the undo signal-value command to restore the bits of the corresponding signaling to the default value. This command applies to E1 voice only. When the router and its peer device, such as a PBX, communicate using the digital E&M signaling, it should be ensured that they interpret the ABCD bits of the received and transmitted idle signaling and seizure signaling in the same way. That is, the signaling of the same type should have the same bit value at both ends. Related command: subscriber-line. Example # Set the ABCD bits of seizure signaling transmitted by the digital E&M subscriber-line are 1011. [H3C-voice-line1/0/0:0] signal-value transmit seize 1011 4.1.42 special-character Syntax special-character character number View R2 CAS view Parameter character: A special character, which can be #, *, A, B, C, or D. number: Code of the register signal, ranging from 11 to 15. Description Use the special-character command to configure the supported special characters during register signal exchange. By default, no special character is configured. This command applies to E1 voice only. 4-45 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands In some countries, besides numerical information, register forward Group I signal of R2 signaling also supports the information containing special characters, such as # and *. To encode these special characters, you need to use the special-character command. Note: z Do not use the special-character command to configure one special character to different signal codes. z Up to six special characters can be configured with register signal codes. z Different special characters must use different signaling values; otherwise, calling may be affected. Example # Configure register signal code of # to 11. [H3C-cas1/0/0:0] special-character # 11 4.1.43 subscriber-line Syntax subscriber-line slot-number: { ts-set-number | 15 } subscriber-line slot-number: 23 View Voice view Parameter slot-number: Indicates the subscriber line number generated in creating Timeslot set or ISDN PRI set, determined by the slot number of the E1V1 module. ts-set-number: Indicates the number of Timeslot set created successfully. 15: Indicates the subscriber line is generated in creating the ISDN PRI set on E1 subscriber line. 23: Indicates the subscriber line is generated in creating the ISDN PRI set on T1 subscriber line. Description Use the subscriber-line command to enter the voice subscriber-line view. This command applies to E1 and T1 voice. 4-46 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands After creating the Timeslot set successfully, the system will generate the subscriber line corresponding to this Timeslot set according to current E1 subscriber-line number and Timeslot set number, and the subscriber line number is “E1 subscriber-line number: Timeslot set number”. After configuring the ISDN PRI set successfully, the system will generate the subscriber line corresponding to this PRI set according to the number of E1 subscriber-line where the current PRI subscriber-line is located, and the subscriber line number is “E1 subscriber-line number: 15”. After configuring the ISDN PRI set successfully on T1 subscriber-line, the system will generate the subscriber line corresponding to this PRI set according to the number of T1 subscriber-line where the current PRI subscriber-line is located, and the subscriber line number is “T1 subscriber-line number: 23”. Related command: timeslots-set, pri-set. Example # Enter the view of subscriber line 1/0/0:5. [H3C-voice] subscriber-line 1/0/0:5 R2 interface encountered [H3C-voice-line1/0/0:5] 4.1.44 timer (digital E&M) Syntax timer { dial-interval | ring-back } seconds timer wait-digit { seconds | infinity } undo timer { dial-interval | ring-back | wait-digit } View Digital E&M voice subscriber-line view Parameter dial-interval seconds: Sets the max waiting time between two digit numbers, it is any integer in the 0 to 300 seconds. By default, the dial-interval seconds is 4 seconds. ring-back seconds: Sets the timeout time for the calling party to wait for the ringback response from the called party. It is any integer in the range 5 to 60000 seconds. By default, the ring-back seconds is 60 seconds. wait-digits second: Sets the timeout time for the called party to wait for the called number. It is any integer in the range 3 to 600 seconds. By default, the wait-digits seconds is 5 seconds. infinity: Indicates that there is no time limit, that is, timeout will not occur. 4-47 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Description Use the timer command to configure the timeout values of the signals in the digital E&M signaling. Use undo timer command to restore them to defaults. This command applies to E1 voice only. The timer ring-back seconds command can be used at the digital E&M subscriber-line to set the timeout time for a calling party to wait for the ringback response from the called party. The timer wait-digit seconds command can be used at the digital E&M subscriber-line to set the timeout time for a called party to wait for the called number. Example # Set the ringback response timer to 30 seconds on a voice subscriber-line. [H3C-voice-line1/0/0:3] timer ring-back 30 4.1.45 timer dtmf (R2) Syntax timer dtmf time undo timer dtmf View R2 CAS view Parameter time: Interval at which R2 signaling sends DTMF signals, in the range 50 to 10000 milliseconds. It defaults to 50 milliseconds. Description Use the timer dtmf command to configure the time interval for sending DTMF signals. Use the undo timer dtmf command to restore the default value. This command applies to E1 voice only. Normally, the originating point sends a DTMF signal upon receiving the line seizure acknowledgement signal. Using this command, you can configure the device at the originating point to send a DTMF signal after the specified time interval, so as to make it appropriate to the number receiving process of the peer PBX, Related command: dtmf { enable | disable }. 4-48 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example # Configure the R2 signaling to send a DTMF signal 800 milliseconds after the reception of the seizure acknowledgement signal. [H3C-cas1/0/0:0] timer dtmf 800 4.1.46 timer register-pulse (R2) Syntax timer register-pulse persistence time undo timer register-pulse View R2 CAS view Parameter persistence time: Duration time of the register pulse signal of R2 signaling, ranging from 100 to 1,000 milliseconds. By default, it is set to 150±30 milliseconds. Description Use the timer pulse command to configure the persistence time of the register pulse signal of R2 signaling (A3, A4 and A6, etc.). Use the undo timer pulse command to restore the default value. This command applies to E1 voice only. When the terminating point sends a backward register pulse signal, such as A3, the signal must persist for a specified time range. When the originating point receives the pulsed A3 signal, it has to send a forward Group II signal. When the originating point recognizes the pulse signal A4, A6, or A15, it stops sending any forward signal, and terminates the register signal exchange. Related command: timer register-complete. Example # Set the persistence time of the register pulse signal of R2 signaling to 300ms. [H3C-cas1/0/0:0] timer register-pulsepersistence 300 4.1.47 timer register-complete (R2) Syntax timer register-complete group-b time undo timer register-complete group-b 4-49 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands View R2 CAS view Parameter group-b time: Timeout value in which the originating point waits for Group-B signal of R2 signaling. The terminating point should send Group-B signal in the specified time when switching to Group-B exchange. It ranges from 100 to 90,000ms. By default, it is set to 30,000ms. Description Use the timer register-complete command to configure the timeout value for register signals of R2 signaling. Use the undo timer register-complete command to restore the default value. This command applies to E1 voice only. Related command: timer dl. Example # Configure the maximum Group-B signal exchange time to 10,000 ms during connecting process. [H3C-cas1/0/0:0] timer register-complete group-b 10000 4.1.48 timer ring (R2) Syntax timer ring { ringback | ringbusy } time undo timer ring { ringback | ringbusy } View R2 CAS view Parameter ringback time: Time interval before ring-back tone is sent, ranging from 1,000 to 90,000ms. By default, it is set to 60,000ms. ringbusy time: Time interval before busy tone is sent, ranging from 1,000 to 90,000ms. By default, it is set to 30,000ms. Description Use the timer ring command to configure the maximum time range before the signal tone of R2 signaling is sent. Use the undo timer ring command to restore it to the default value. 4-50 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands This command applies to E1 voice only. By manually configuring the sending command, you can enable the device at the originating point to send ring-back and busy tones to the calling party. The timer ring command helps you identify these signals. Related command: sendring. Example # Configure the timeout interval for R2 signaling ring-back tone to 10,000ms. [H3C-cas1/0/0:0] timer ring ringback 10000 4.1.49 timer dl (R2) Syntax timer dl { answer | clear-back | clear-forward | seizure | re-answer | release-guard } time undo timer dl { answer | clear-back | clear-forward | seizure | re-answer | release-guard } View R2 CAS view Parameter answer time: Timeout interval during which R2 waits for an answer signal. The terminating point should send back an answer signal in the specified time after a seizure acknowledgement signal is sent. It ranges from 10 to 120000 milliseconds and defaults to 60000 milliseconds. clear-back time: Timeout interval of R2 clear-back signal. After it sends a clear-back signal, the terminating point should recognize the forward signal sent by the originating point in the specified time. It ranges from 100 to 60000 milliseconds. By default, it is set to 10000 milliseconds. clear-forward time: Timeout interval of R2 clear-forward signal. After the originating point sends a clear-forward signal, the terminating point should send back a corresponding line signal in the specified time, such as clear-back signal or release guard signal. It ranges from 100 to 60000 milliseconds. By default, it is set to 10000 milliseconds. seizure time : Timeout value of R2 seizure signal. After the originating point sends a seizure signal, the terminating point should send back a seizure acknowledgement signal in the specified time. It ranges from 100 to 5000 milliseconds. By default, it is set to 1000 milliseconds. 4-51 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands re-answer time: Timeout interval of R2 re-answer signal. When the originating point recognizes the clear-back signal, if the terminating point does not send a re-answer signal in the specified time, the originating point releases the line. It ranges from 100 to 60000 milliseconds. By default, it is set to 1000 milliseconds. release-guard time: Timeout interval of R2 release guard signal. When the originating point sends a clear-forward signal, the terminating point should send a release guard signal in the specified time after it sends back a clear-back signal. It ranges from 100 to 60000 milliseconds. By default, it is set to 10000 milliseconds. Description Use the timer dl command to configure the timeout value of line signals of R2 signaling. Use the undo timer dl command to restore the default value. This command applies to E1 voice only. Related command: timer complete. Example # Configure the timeout interval of R2 signaling seizure signal to 300ms. [H3C-cas1/0/0:0] timer dl seize 300 4.1.50 timeslot-set Syntax timeslot-set ts-set-number timeslot-list timeslots-list signal { e&m-delay | e&m-immediate | e&m-wink | r2 } undo timeslot-set ts-set-number View CE1/PRI interface view Parameter ts-set-number: Specifies the identification number of a certain Timeslot set, with the value ranging from 0 to 30. timeslots-list: Specifies a timeslot range. The expression includes the single digit, two digits separated by “,”, a pair of digits separated by “-”, or the combination form (e.g., 1-14, 15, 17-31). The value range of digits is the integer from 1 to 31. signal: Specifies the binding signaling mode of this Timeslot set, which is generally used to configure the signaling mode adopted by the central office. It includes the following types of signaling: z e&m-delay: Adopts the delay-start mode in the digital E&M signaling. 4-52 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands z e&m-immediate: Adopts the immediate-start mode in the digital E&M signaling. z e&m-wink: Adopts the wink-start mode in the digital E&M signaling. z r2: Specifies that the signaling mode adopt ITU-T Q.421 digital line signaling R2, which is the most common configuration signaling. Description Use the timeslot-set command to configure the timeslot set to perform R2 signaling and digital E&M signaling configurations. Use the undo timeslot-set command to cancel the specified timeslot set. By default, no TS set is configured, and if the digital E&M signaling is used, then adopts e&m-immediate mode. This command applies to E1 voice only. The Timeslot set is actually the logical subscriber line abstracted from the physical E1 interface, mainly used for configuring R2 signaling, digital E&M signaling and other voice functions. In one E1 interface only one Timeslot set can be defined. In defining the timeslot range of Timeslot set, the timeslot range can be distributed is 1-15 and 17-31, and the No.16 timeslot is reserved as the transmission channel of the out-of-band signaling. When using the digital E&M signaling delay-start mode (e&m-delay), If the calling side off-hooks to occupy the trunk line, the connected peer (e.g., PBX) will also enter the off-hook state to answer the calling party and will remain in that state until it is ready for receiving the address message. In this case, the PBX enters the hook-up state (this interval is the delay-dial period). The calling party sends the address message, and PBX connects this call to the called party, and thus the two parties can begin their communications. When the digital E&M signaling e&m-immediate mode is adopted, the calling party off-hooks to wait for the confirmation of time, and then it sends the dialing address message to the connected peer such as a PBX. During the process, it does not detect whether the PBX is ready to receive. When the digital E&M signaling e&m-wink mode is adopted, the calling party first off-hooks to occupy the trunk line while the connected peer (e.g., a PBX) remains in the hook-up state until it receive the connecting signal from the calling party. In this case, the wink signal sent by the PBX indicates that it has been ready. Upon receiving the wink signal, the calling party begins to send the address message and the PBX will connect the call to the called party and thus the two parties can begin their communications. Only after establishing TS set successfully can the command subscriber-line be used to enter the subscriber line and configure the voice-related attributes. Related command: subscriber-line, cas. 4-53 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example # Create a timeslot set numbered 5, including 1 to 31 timeslots and using R2 signaling. [H3C-E1 1/0/0] timeslot-set 5 timeslot-list 1-31 signal r2 4.1.51 trunk-direction Syntax trunk-direction timeslots timeslots-list { in | out | dual } undo trunk-direction timeslots timeslots-list View R2 CAS view Parameter timeslots-list: Specifies the range of trunk timeslot. The expression includes the single digit, two digits separated by “,”, a pair of digits separated by “-”, or the combination form (e.g., 1-14, 15, 17-31). The value range of digits is the integer from 1 to 31. in: Indicates the trunk is the incoming trunk. out: Indicates the trunk is the outgoing trunk. dual: Indicates the trunk is the bidirectional trunk. Description Use the trunk-direction command to configure the E1 trunk direction. Use the undo trunk-direction command to restore the default value. By default, configure the bidirectional trunk. This command applies to E1 voice only. When configuring the E1 trunk direction as incoming trunk, this trunk will not load any outgoing call. When configuring the E1 trunk direction as outgoing trunk, this trunk can only be used for the outgoing call and not for incoming call. When configuring it as bidirectional trunk, it can load the outgoing call and incoming call respectively according to the initiative of originating call. To keep the E1 communication appropriate, if the E1 trunk adopts the mode of incoming trunk or outgoing trunk, then one end of it must be ensured as incoming and other as outgoing, or the connection will fail. If both ends of E1 trunk adopt the bidirection mode, it is required to use the command of select-mode to adjust the trunk selection policy to avoid the simultaneous hold of timeslot by the two sides of communication. In configuration avoid that one end is bidirectional trunk while the other end is outgoing trunk, or the call from the end configured as the bidirectional trunk will always fail. 4-54 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Related command: cas, select-mode. Example # Configure the trunk direction as bidirectional trunk for timeslot set 5 on E1 port 1/0/0. [H3C-cas1/0/0:5] trunk-direction timeslots 1-31 dual 4.1.52 ts Syntax ts { block | open | query | reset } timeslots timeslots-list View R2 CAS view Parameter block: Indicates blocking the trunk circuit of the specified timeslot. open: Indicates opening the trunk circuit of the specified timeslot. query: Indicates querying the trunk circuit of the specified timeslot. reset: Indicates resetting the trunk circuit of the specified timeslot. timeslots timeslots-list: Specifies a timeslot range. The expression includes the single digit, two digits separated by “,”, a pair of digits separated by “-”, or the combination form (e.g., 1-14, 15, 17-31). The value range of digits is the integer from 1 to 31. Description Use the ts command to maintain the trunk circuit of the specified timeslot. This command applies to E1 voice only. To block the trunk circuit means that the circuit no longer loads the service information, that is, this circuit is set manually as unavailable. To open the trunk circuit is the inverse process of the blocking operation, which can reset the trunk circuit as available and enable it to load service information. To query the trunk circuit will display the busy/idle, opened/blocked status of the circuit in a real time way. To reset the trunk circuit refers to re-initializing the state of trunk circuit. Generally, if the circuit state cannot be restored to normal in blocking or opening the circuit manually, it is required to perform the resetting. If the circuit cannot be reset automatically and correctly because of other reasons,, it is generally required to reset manually the circuit, too. Related command: cas, mfc. 4-55 Command Manual – Voice Comware V3 Chapter 4 E1/T1 Voice Configuration Commands Example # Reset the circuit of timeslots 1-15 in No.5 Timeslot set and query the status of the circuit of timeslots 1-31. [H3C-cas1/0/0:5] ts reset timeslots 1-15 [H3C-cas1/0/0:5] ts query timeslots 1-31 4-56 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands Chapter 5 Fax Configuration Commands 5.1 Fax Configuration Commands 5.1.1 cngced-detection Syntax cngced-detection threshold times undo cngced-detection View Voice subscriber line view Parameter threshold: Calling tone/Called terminal identification (CNG/CED) detection threshold with fax. It is an integer in the range 0 to 30 and its default is 0. times: Threshold of CNG/CED detections with fax. It is an integer in the range 0 to 100 and the default is 10. Description Use the cngced-detection command to set the threshold parameters for CNG/CED signal detection. Use the undo cngced-detection command to restore the default. CNG is generated when the fax machine at the calling end starts, while CED is generated when the fax machine at the called end starts. A voice gateway determines fax state by detecting CNG/CED. The likelihood of detection failure or error exists however depending on the environment where the device is deployed. As a solution, you can use this command to tune the CNG/CED detection sensitivity and reliability. As the threshold value is increasing, detection reliability increases; if you assign a value that is too large, however, detection may fail as well. The same applies to the setting of the times argument. The times argument specifies the lower limit of the CNG/CED duration. For instance, the default of times is 10; this means the CNG/CED signal must last at least 300 milliseconds to be regarded valid. As the times value is incrementing by one, the minimum CNG/CED signal duration increases by 30 milliseconds. 5-1 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands Example # Set the threshold argument to 5 and the times argument to 20 in voice subscriber-line view for CNG/CED signal detection. [H3C-voice-line0] cngced-detection 5 20 5.1.2 debugging voice fax Syntax debugging voice fax { all | api | cc | channel [ channel-no ] | nonstandard-compatible | motorola-compatible |controller | error-all | frf11 |ipp | transform | vofr | t38 | comware-relay } View User view Parameter all: Enables all fax debugging. api: Enables API function debugging of fax. cc: Enables main task debugging of fax. channel [ channelno ]: Enables debugging of the specified channel. nonstandard-compatible: Enables debugging of the nonstandard-compatible fax protocol. motorola-compatible: Enables debugging of the motorola-compatible fax protocol. controller: Enables fax controller debugging. error-all: Enables fax error debugging at all levels. frf11: Enables frf11 debugging. ipp: Enables debugging of the messages between the fax and IPP modules. transform: Enables FAX and GS transform debugging. vofr: Enables debugging between FAX and VOFR. comware-relay: Enables Standard-FaxRelay debugging of fax. t38: Enables Fax T38 debugging. Description Use the debugging voice fax command to enable fax debugging. The debugging voice fax channel channel-number command is independent of the debugging voice fax all command. The debugging voice fax channel channel-number command enables debugging for only the specified channel; it does not enable debugging for the specific items. To view the debugging information of a 5-2 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands specific item, you must enable debugging for it with the corresponding command. The debugging voice fax all command enables debugging for all items except for channel. The undo debugging voice fax all command are not completely independent of the undo debugging voice fax channel command. The undo debugging voice fax all command also disables debugging for channel, while the undo debugging voice fax channel command disables debugging only for channel. Example # Enables main task debugging of fax. debugging voice fax cc 5.1.3 default entity fax Syntax default entity fax baudrate { 2400 | 4800 | 9600 | 14400 | disable | voice } default entity fax ecm default entity fax level level default entity fax local-train threshold value default entity fax nsf-on default entity fax protocol { standard-t38 | t38 | nonstandard } [ hb-redundancy number | lb-redundancy number ] default entity fax protocol { pcm { g711alaw | g711ulaw } { private | standard } } default entity fax protocol none default entity fax support-mode { rtp | sip-udp | vt } default entity fax train-mode { local | ppp } undo default entity fax { baudrate | ecm | level | local-train threshold | nsf-on | protocol | support-mode | train-mode } View Voice dial program view Parameter baudrate: Specifies the maximum transmission speed of the fax. It defaults to voice. z 2400: Sets the maximum transmission speed to 2400 bps. z 4800: Sets the maximum transmission speed to 4800 bps. z 9600: Sets the maximum transmission speed to 9600 bps. z 14400: Sets the maximum transmission speed to 14400 bps. z disable: Disables the fax forwarding faculty. z voice: Sets the fax speed to the allowed maximum voice speed. 5-3 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands ecm: Enables fax error correction mode. It is disabled by default. level level: The level of the fax transmission signal. It defaults to 15. local-train: Specifies the threshold of fax local training. z threshold value: Specifies the threshold of fax local training. It defaults to 10. nsf-on: Enables NSF message transmission. It is disabled by default. protocol: Specifies the transport protocol of the fax. By default, the T.38 fax protocol applies. Both hb-redundancy number and lb-redundancy number default to 0. z standard-t38: Adopts the standard T38 (UDP) fax protocol, which supports H.323-T38 and SIP-T38 protocols. z nonstandard: Adopts the nonstandard protocol compatible with private fax protocols. z pcm: Enables PCM mode. z g711alaw: Adopts G.711 a-law. z g711ulaw: Adopts G.711 μ-law. z t38: Adopts the T.38 fax protocol. z hb-redundancy number: Number of redundant high-speed T.38 packets. z lb-redundancy number: Number of redundant low-speed T.38 packets. pcm: Enables PCM fax mode. z g711alaw: Adopts G.711 a-law. z g711ulaw: Adopts G.711 μ-law. private: Private PCM fax negotiation mode owned by our company. standard: Standard PCM fax negotiation mode in the industry. By default, the global voice entity fax protocol is the private T.38 owned by our company. none: Disables the fax function. support-mode: Fax transport format. It defaults to real time protocol (RTP). z rtp: Adopts RTP fax mode. z sip-udp: Uses the fax packets without RTP header. z vt: Adopts VT fax mode. train-mode: Specifies the fax training mode. It defaults to end to end mode or PPP. local: Adopts local training. ppp: Adopts end to end training. Description Use the default entity fax command to set the default fax parameter settings globally. Use the undo default entity fax command to restore the system-default default settings. 5-4 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands Note: When G.711 voice encoding/decoding recommendation is configured for the two parties, the fax baudrate disable command functions the same as the default entity fax protocol pcm command. Example # Set the maximum global fax speed to 9600 bps. [H3C-voice-dial] default entity fax baudrate 9600 # Configure the fax energy level to 20 globally. [H3C-voice-dial] default entity fax level 20 # Globally configure the fax protocol for the voice entity to PCM coding/decoding. system-view [H3C] voice-setup [H3C-voice] dial-program [H3C-voice-dial] default entity fax protocol pcm g711alaw standard 5.1.4 default entity modem compatible-param Syntax default entity modem compatible-param integer undo default entity modem compatible-param View Voice dial program view Parameter integer: Compatible parameter value. It defaults to 100. Description Use the default entity modem compatible-param command in the case that you fail to use standard Modem PCM to interoperate with devices from other vendors. Use the undo default entity modem compatible-param command to restore the voice entity Modem compatible parameter globally to the default. By default, the Modem compatible parameter for global voice entity is 100. Example # Configure the Modem compatible parameter for the global voice entity to 98. 5-5 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands system-view [H3C] voice-setup [H3C-voice] dial-program [H3C-voice-dial] default entity modem compatible-param 98 5.1.5 default entity modem protocol Syntax default entity modem protocol { none | pcm { g711alaw | g711ulaw } { private | standard } } undo default entity modem protocol View Voice dial program view Parameter none: Disables modem. pcm: Enables PCM fax mode. z g711alaw: Adopts PCM G.711 a-law. z g711ulaw: Adopts PCM G.711 μ-law. private: Private PCM Modem negotiation mode owned by our company. standard: Standard PCM Modem negotiation mode in the industry. Description Use the default entity modem protocol none command to globally disable Modem function of voice entity. Use the default entity modem protocol pcm command to globally configure PCM Modem protocol type of voice entity, including coding/decoding and Modem negotiation mode. Use the undo default entity modem protocol command to restore the Modem protocol to default value. By default, Modem protocol for global voice entity is none, that is, the Modem function is disabled. Example # Globally configure the Modem protocol for voice entity to PCM coding/decoding. system-view [H3C] voice-setup [H3C-voice] dial-program [H3C-voice-dial] default entity modem protocol pcm g711alaw standard 5-6 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands # Disable the Modem function of global voice entity. system-view [H3C] voice-setup [H3C-voice] dial-program [H3C-voice-dial] default entity modem protocol none 5.1.6 display voice fax Syntax display voice fax { statistics | trans-statistics } View Any view Parameter statistics: Displays the fax statistics of the fax module. trans-statistics: Displays the statistics in the fax transformation module. Description Use the display voice fax command to view the fax statistics in the fax module and the statistics in the fax transformation module as well. Example # Display the statistics about the fax module. [H3C] display voice fax statistic # Display the statistics about the fax transformation module. [H3C] display voice fax trans-statistics Statistic of FOFR to FOIP { Create_FaxTransformNode_Times: 0 Send_Data_Packets: 0 Recieve_Data_Packets: 0 Recieve_InvalidData_Packets: 0 V21Data_WithOutRedundance_Packets: 0 V21Data_WithRedundance_Packets: 0 T4Data_WithOutRedundance_Packets: 0 T4Data_WithRedundance_Packets: 0 Fax_ECM_Times: 0 Fax_NoECM_Times: 0 } Statistic of FOIP to FOFR 5-7 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands { Create_FaxTransformNode_Times: 0 Send_Data_Packets: 0 Recieve_Data_Packets: 0 Recieve_InvalidData_Packets: 0 V21Data_WithOutRedundance_Packets: 0 V21Data_WithRedundance_Packets: 0 T4Data_WithOutRedundance_Packets: 0 T4Data_WithRedundance_Packets: 0 } Table 5-1 Description on the transformation statistics Field Description Create_FaxTransformNode_Times Times of transform node created from IP fax side to FR fax side Send_Data_Packets Number of packets sent from IP fax side to FR fax side Recieve_Data_Packets Number of packets received at IP fax side from FR fax side Recieve_InvalidData_Packets Number of invalid packets received at IP fax side from FR fax side V21Data_WithOutRedundance_Packets Number of packets sent from IP fax side to FR fax side without redundancy V.21 data V21Data_WithRedundance_Packets Number of packets sent from IP fax side to FR fax side with redundancy V.21 data T4Data_WithOutRedundance_Packets Number of packets sent from IP fax side to FR fax side without redundancy T.4 data T4Data_WithRedundance_Packets Number of packets sent from IP fax side to FR fax side with redundancy T.4 data Fax_ECM_Times Times of fax communication using ECM at IP fax side Fax_NoECM_Times Times of fax communication using non-ECM at IP fax side Create_FaxTransformNode_Times Times of transform node created from FR fax side to IP fax side Send_Data_Packets Number of packets sent from FR fax side to IP fax side Recieve_Data_Packets Number of packets received at FR fax side from IP fax side 5-8 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands Field Description Recieve_InvalidData_Packets Number of invalid packets received at FR fax side from IP fax side V21Data_WithOutRedundance_Packets Number of packets sent from FR fax side to IP fax side without redundancy V.21 data V21Data_WithRedundance_Packets Number of packets sent from FR fax side to IP fax side with redundancy V.21 data T4Data_WithOutRedundance_Packets Number of packets sent from FR fax side to IP fax side without redundancy T.4 data T4Data_WithRedundance_Packets Number of packets sent from FR fax side to IP fax side with redundancy T.4 data 5.1.7 fax baudrate Syntax fax baudrate { 14400 | 2400 | 4800 | 9600 | disable | voice } View Voice entity view Parameter 14400: Negotiates first according to V.17 fax protocol, the highest fax baudrate is 14400bit/s. 2400: The first fax baudrate is 2400bps. 4800: Negotiates first according to V.27 fax protocol, the highest fax baudrate is 4800bps. 9600: Negotiates first according to V.29 fax protocol which has the priority, the highest fax baudrate is 9600bps. disable: Disables fax function. voice: Decides the highest rate enabled by fax first according to the different voice encoding/decoding protocols. By default, the parameter of voice is adopted to decide the max fax baudrate. Description Use the fax baudrate command to configure the highest fax baudrate enabled by the gateway. 5-9 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands If the rate is set to be others rather than “disable” and “voice”, rate negotiation is performed first according to the fax protocol which is correspondent to the rate. Here the rate is the highest rate enabled but not specified one. When the rate is set as voice, decide the highest rate enabled by fax first according to the difference of voice encoding/decoding protocols. z If G.711 voice encoding/decoding protocol is used, the fax baudrate is 14400bit/s and the corresponding fax protocol is V.17. z If G.723.1 Annex A voice encoding/decoding protocol is used, the fax baudrate is 4800bit/s and the corresponding fax protocol is V.27. z If G.726 voice encoding/decoding protocol is used, the fax baudrate is 14400 bit/s, and the corresponding fax protocol is V.17. z If G.729 voice encoding/decoding protocol is used, the fax baudrate is 9600bit/s and the corresponding fax protocol is V.29. If the rate is set as “disable”, fax function is disabled. Note: When G.711 voice encoding/decoding protocol is configured for the two parties, the fax baudrate disable command functions the same as the default entity fax protocol pcm command. Example # Set the gateway to use V.29 fax protocol to perform rate negotiation. [H3C-voice-dial-entity1] fax baudrate 9600 5.1.8 fax ecm Syntax fax ecm undo fax ecm View Voice entity view Parameter None 5-10 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands Description Use the fax ecm command to configure the forced adoption of ECM mode at the gateway, that is, to make the facsimiles at both ends support ECM mode. Use the undo fax ecm command to cancel the ECM mode at the gateway. By default, the ECM mode is not used on the gateway. This command is used to perform the forced limitation at the gateway. If the facsimile terminals at both ends support ECM mode, but the non-ECM mode has been configured at the gateway, then the non-ECM mode is selected. If one or two of the facsimile terminals at both ends do not support the ECM mode, the non-ECM mode is selected. Only when the facsimile terminals at both ends support the ECM mode, and the gateway doesn’t perform the forced limitation to enable the non-ECM mode, is the ECM mode selected. You must enable ECM mode for the POTS and VoIP entities corresponding to the fax sender and receiver in ECM mode. Example # Configure forced adoption of ECM mode at the gateway. [H3C-voice-dial-entity1] fax ecm 5.1.9 fax level Syntax fax level level undo fax level View Voice entity view Parameter level: Gateway carrier transmitting energy level, namely the transmit energy level attenuation value, ranging from 3 to 60 dBm. By default, it is 15, meaning a carrier transmit energy level of 15 dBm. The greater the level value, the higher the energy. The smaller the level value, the greater the attenuation. Description Use the fax level command to configure the gateway carrier transmitting energy level. Use the undo fax level command to restore it to the default value. Usually the default of the gateway carrier transmitting energy level is acceptable. If fax still can’t be sent when other configurations are correct to try to adjust the gateway carrier transmitting energy level. The smaller the level is, the greater the energy is. 5-11 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands Example # Configure the gateway carrier transmitting energy level as 20. [H3C-voice-dial-entity1] fax level 20 5.1.10 fax nsf-on Syntax fax nsf-on undo fax nsf-on View Voice entity view Parameter None Description Use the fax nsf-on common to configure the fax faculty transmission mode as Not Standard mode. Use the undo nsf-on command to restore the default fax faculty transmission mode. By default, the fax faculty transmission mode is undo nsf-on. This command is only applied to IP Fax. In some occasions like communicating with encrypted fax, the Not Standard Faculty, or NSF for short, is rather important for fax communication. It is necessary to configure the fax nsf-on command, so that the fax terminals on both sides will first exchange NSF message at the beginning of transmission, and then complete the follow-up fax negotiation according to NSF and communicate with each other. NSF messages are T.30 compliant and carry private information. To allow NSF field to be carried during fax faculty signal transmission, the following conditions must be met: z Fax terminals must support NSF transmission mode. z The fax faculty signal transmission mode must be set to Not Standard in the POTS and VoIP entities for the participating fax terminals. Example # Configure to perform fax transmission with NSF. [H3C-voice-dial-entity1] fax nsf-on 5-12 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands 5.1.11 fax protocol Syntax fax protocol { t38 | standard-t38 | nonstandard } [ lb-redundancy number | hb-redundancy number ] fax protocol { pcm { g711alaw | g711ulaw } { private | standard } } undo fax protocol View Voice entity view Parameter t38: Uses T.38 fax protocol. With this protocol, fax setup speed is high. standard-t38: Uses the standard T38 negotiation mode specified by H323 or SIP to fax lb-redundancy number: Configures the number of low-speed redundant packets. number ranges from 0 to 5, and default value is 0. hb-redundancy number: Configures the number of high-speed redundant packets. number ranges from 0 to 2, and default value is 0. nonstandard: Uses the fax protocol compatible with the industry-leading devices. pcm: Uses PCM coding/decoding. z g711alaw: Enables PCM G.711A-law voice coding/decoding. z g711ulaw: Enables PCM G.711 μ-law voice coding/decoding. private: Private PCM Modem negotiation mode owned by our company. standard: Standard PCM Modem negotiation mode in the industry. Description Use the fax protocol command to configure the protocol for intercommunication with other devices or enable the fax Passthrough mode, and configure the number of redundant packets sent via the T.38 fax protocol. Use the undo fax protocol command to restore the default number of redundant packets and disable the fax Passthrough. The argument standard-t38 indicates the standard T38 negotiation mode specified by H323 or SIP adopted. By default, fax Passthrough mode is disabled and fax protocol for single voice entity is that for global voice entity. This command is only applied to IP Fax. Low-speed data refers to the V.21 command data, and high-speed data refers to the TCF and graphic data. 5-13 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands To intercommunicate with the mainstream fax terminals of the industry, the mainstream fax protocol must be used. Likewise, to intercommunicate with other fax terminals supporting the T.38 protocol, the T.38 protocol must be adopted. As the mainstream devices do not support local training mode for fax, the end-to-end training mode must be adopted in order to implement intercommunication with the mainstream devices of the industry. Increasing the number of redundant packets will improve reliability of network transmission and reduce packet loss ratio. A great amount of redundant packets, however, can increase bandwidth consumption 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 properly according to the network bandwidth. The fax Passthrough is easily influenced by such factors as loss of packet, jitter and delay, so the clock on both sides of the communication must be kept synchronous. At present, only G.711Alaw and G.711μlaw compression methods are supported, and the voice activity detection (VAD) function should be disabled for fax Passthrough. Example # Configure the number of high-speed redundant packets sent through the T.38 fax recommendation as 2. [H3C-voice-dial-entity1] fax protocol t38 hb-redundancy 2 # Configure fax protocol for voice entity 100 to PCM coding/decoding. system-view [H3C] voice-setup [H3C-voice] dial-program [H3C-voice-dial] entity 100 [H3C-voice-dial-entity100] fax protocol pcm g711alaw standard 5.1.12 fax support-mode Syntax fax support-mode { rtp | vt | sip-udp } undo fax support-mode View Voice entity view Parameter rtp: RTP mode. vt: VT mode, the mode when interworking with the VocalTec gateway. sip-udp: Specifies SIP UDP mode. 5-14 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands Description Use the fax support-mode command to configure the fax interworking mode with other equipments. Use the undo fax support-mode command to restore the fax interworking mode with other equipment to its default. By default, the RTP mode is adopted. This command is only applied to IP Fax. Under common conditions, use the rtp mode. When interworking with the VocalTec gateway, the vt mode is adopted. Example # Configure the mode when interworking with VocalTec device to vt. [H3C-voice-dial-entity1] fax support-mode vt 5.1.13 fax train-mode Syntax fax train-mode { local | ppp } undo fax train-mode View Voice entity view Parameter local: Uses the local training mode. ppp: Uses the end-to-end training mode. Description Use the fax train-mode command to configure the training mode used by the gateway. Use the undo fax train-mode command to restore the training mode used by the gateway to the default. By default, the ppp mode is adopted. The local training mode means that the gateway participates in the rate training between the facsimile terminals at both ends. In this mode, first carry on the training between the facsimile terminals and between the gateways respectively, then the receiving gateway will transmit the training result of the receiving end to the gateway at the transmitting end. The transmitting gateway will decide the final packet transmission rate according to the training results of both ends. 5-15 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands The end-to-end training mode means that the gateway does not participates in the rate training between the facsimile terminals at both ends. In this mode, the rate training is carried on between the facsimile terminals at both ends, and it is transparent to the gateway. Example # Set the mode used by the gateway as the local training mode. [H3C-voice-dial-entity4] fax train-mode local 5.1.14 modem compatible-param Syntax modem compatible-param integer undo modem compatible-param View Voice entity view parameter integer: Compatible parameter value. It defaults to 100. Description Use the modem compatible-param command in the case that you fail to use standard Modem PCM to interoperate with devices from other vendors. Use the undo default entity modem compatible-param command to restore the voice entity Modem compatible parameter globally to the default. By default, the Modem compatible parameter for global voice entity is 100. Example # Configure the Modem compatible parameter of voice entity 100 to 98. system-view [H3C] voice-setup [H3C-voice] dial-program [H3C-voice-dial] entity 100 [H3C-voice-dial-entity100] modem compatible-param 98 5.1.15 modem protocol Syntax modem protocol { none | pcm { g711alaw | g711ulaw } { private | standard } } undo modem protocol 5-16 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands View Voice entity view parameter none: Disables modem. pcm: Enables PCM fax mode. z g711alaw: Adopts PCM G.711 a-law. z g711ulaw: Adopts PCM G.711 μ-law. private: Private PCM Modem negotiation mode owned by our company. standard: Standard PCM Modem negotiation mode in the industry. Description Use the modem protocol none protocol to disable Modem function for single voice entity. Use the modem protocol pcm command to configure PCM Modem protocol type for single voice entity, including coding/decoding and Modem negotiation mode. Use the undo modem protocol command to restore the Modem protocol for single voice entity to the default value. By default, Modem protocol for single voice entity is that for global voice entity. Example # Configure the Modem protocol of voice entity 100 to PCM coding/decoding. system-view [H3C] voice-setup [H3C-voice] dial-program [H3C-voice-dial] entity 100 [H3C-voice-dial-entity100] modem protocol pcm g711alaw standard # Disable the Modem function of voice entity. system-view system-view [H3C] voice-setup [H3C-voice] dial-program [H3C-voice-dial] entity 100 [H3C-voice-dial-entity100] modem protocol none 5.1.16 reset voice fax statistics Syntax reset voice fax statistics 5-17 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands View User view Parameter None Description Use the reset voice fax statistics command to reset fax statistics. Example # Clear fax statistics. reset voice fax statistics 5.1.17 reset voice fax trans-statistics Syntax reset voice fax trans-statistics View User view Parameter None Description Use the reset voice fax trans-statistics command to reset the statistics for the FAX transform module. Example # Clear the statistics for the fax transform module. reset voice fax trans-statistics 5.1.18 voip h323-conf tcs-t38 Syntax voip h323-conf tcs-t38 undo voip h323-conf tcs-t38 View Voice view 5-18 Command Manual – Voice Comware V3 Chapter 5 Fax Configuration Commands Parameter None Description Use the voip h323-conf tcs-t38 command to enable the voice gateway to include the T.38 capability description in its capability set when it is in H.323 slow-start mode. Use the undo voip h323-conf tcs-t38 command to disable the voice gateway to include the T.38 capability description in its capability set when it is in H.323 slow-start mode. By default, T.38 description is included. As NetMeeting does not support T.38 capability description parsing, you must disable the voice gateway in H.323 slow-start mode to include the T.38 capability description in its capability set in order to work with NetMeeting. Note: As this command is globally effective, the configuration of its undo form can disable all the voice entities to carry the T.38 capability description in their capability sets. If interoperability with NetMeeting is required only by a voice entity, you can disable fax using the fax baudrate disable command or set the fax mode to a non-T.38 mode, pcm, or nonstandard for example. Example # Disable the voice gateway to carry T.38 capacity description in its capacity set when it is in H.323 slow-start mode. [H3C-voice] undo voip h323-conf tcs-t38 5-19 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Chapter 6 Voice RADIUS Configuration Commands 6.1 Voice RADIUS Configuration Commands 6.1.1 aaa-client Syntax aaa-client View Voice view Parameter None Description Use the aaa-client command to enter Voice AAA view. Use the quit command to exit this view. Related command: accounting, authentication-did, clienttype, local-user. Example # Enter Voice AAA view. [H3C] voice-setup [H3C-voice] aaa-client [H3C-voice-aaa] 6.1.2 authentication Syntax authentication undo authentication View Voice access-number view Parameter None 6-1 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Description Use the authentication command to enable user authentication and authorization for the access service number. Use the undo authentication command to disable user authentication and authorization for the access service number. By default, user authentication and authorization is disabled for all access service numbers. If user authentication and authorization is enabled for an access service number, the user who uses this access service number can be authorized and dial the IP phone only after it has been authenticated. If user authentication and authorization is disabled, the user who uses this access service number can directly dial the IP phone without being authenticated. Related command: gw-access-number, accounting optional, authentication-did. Example # Enable user authentication and authorization for the two-stage dial access service number 2005. [H3C-voice-dial-anum2005] authentication 6.1.3 authentication-did Syntax authentication-did undo authentication-did View Voice AAA view Parameter None Description Use the authentication-did command to enable authentication for all one-stage dial (direct dial) users. Use the undo authentication-did command to disable authentication for one-stage dial users. By default, authentication is disabled for one-stage dial users. This command is available to one-stage dial (that is, dialing the called number directly) users but not to two-stage dial (that is, dialing the access service number) users. 6-2 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands After this command is configured, the calling number in a call from a one-stage dial user is extracted and sent to the RADIUS Server for authentication. Only after passing authentication can the user obtain services. If the authentication fails, the user is disconnected and cannot make the IP phone call. If authentication is disabled for one-stage dial users, one-stage dial users can directly dial IP phones without being authenticated. For related commands, see accounting. Example # Enable authentication for one-stage dial users. [H3C-voice-aaa] authentication-did 6.1.4 callednumber receive-method Syntax callednumber receive-method { immediate | terminator } undo callednumber receive-method View Voice access-number view Parameter immediate: Places a call immediately after all digits of the called number are collected. terminator: Places a call only after a pound sign (#), the terminator, is received. Description Use the callednumber receive-method command to enable the device to place a call immediately after all digits of the called number are collected or after a pound sign (#), the terminator, is received. Use the undo callednumber receive-method command to restore the default. By default, the terminator keyword applies. Normally, in a two-stage dial procedure, the user needs to input a pound sign (#) to indicate the end of a called number. To free the user from the trouble of inputting the terminator, you may configure the callednumber receive-method immediate command allowing the router to place a call immediately after all digits of the called number are collected. Example # Set the called number receive-method to immediate for access number 17990. [H3C-voice-dial-anum17990] callednumber receive-method immediate 6-3 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands 6.1.5 card-digit Syntax card-digit card-digit View Voice access-number view Parameter card-digit: The card digits, in the range from 1 to 31. By default, the user card number is in 12 digits. Description Use the card-digit command to configure the number of digits in a card number for an access service number. This command is used to configure digits of the card number of the two-stage dialing card number process (use the card number/password for ID authentication) user. Once the digits are specified, all users using this access service number must enter the card number in specified digits. If the dialing process of the access service number to be configured is not specified as the card number process through the process-config command, this command will not take effect. Only after the dialing process is set as the card number process, can this command function properly in the Voice access-number view. Related command: gw-access-number, process-config, password-digit. Example # Set the user card number in 10 digits for the access service number 18901. [H3C-voice-dial-anum18901] card-digit 10 6.1.6 cdr Syntax cdr { buffer [ size-number ] | duration [ timer-number ] | threshold { threshold | default } } undo cdr View Voice AAA view Parameter buffer: Saves the call records by setting the upper limit of the number of records. 6-4 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands size-number: Number of the saved call detail records, in the range from 0 to 500, with the value 0 indicating that no call detail record is saved. In max-size mode, the default value of size-number is 50. duration: Saves the call records by setting the maximum retaining time of records (start from the moment when a conversation ends). timer-number: Retaining time of call detail records, in minutes, with the value range from 0 to 2147483647, and the value 0 indicating that no call detail record is saved. In max-size mode, the default value of timer-number is 15. threshold threshold: Threshold for call history output, in the range 0 to 100. It defaults to 80. default: Default threshold for call history output. Description Use the cdr command to configure the saving rule for call detail record. Use the undo cdr command to restore the default saving rule. This command is used to set the saving rule for call detail records. The system will save certain amount of call detail record information according to the rule set by the user. Related command: display aaa unsent-h323-call-record, display voice call-history-record. Note: In Comware (Versatile Routing Platform), at most 500 call detail records can be saved. That is, even if it is specified that call detail records should be saved according to the maximum retaining time, no more than 500 records can be saved in the system. If a large amount of traffic occurs in a certain period of time and more than 500 call detail records to be stored are generated which satisfy the time requirement, the excessive records that end earlier will be deleted, though they comply with the saving rule. Example # Specify that at most 400 call detail records can be saved. [H3C-voice-aaa] cdr buffer 400 # Specify that the detail records of the conversation ending within 10 minutes should be saved. [H3C-voice-aaa] cdr duration 10 6-5 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands 6.1.7 delay receive-dial Syntax delay receive-dial delay-time undo delay receive-dial View Access number view Parameter delay-time: Delay in milliseconds, in the range of 100 to 5,000 with a default of 300. Description Use the delay receive-dial command to configure the delay in collecting the digits of a called number in two-stage dialing. Use the undo delay receive-dial command to restore the default delay. By default, no delay is configured. In this case, if there is too long an interval before the last digit of an access number is dialed, the system will mistakenly consider this digit as one of the called number in the second stage. If the delay is too small, the system may fail to collect the digits of the called number completely when there is too small an interval between the access number and the first digit of the called number. Therefore, the delay should be configured according to the practical conditions. Example # Specify the access number as 2005 and set the delay in collecting digits of a called number in two-stage dialing to 500 milliseconds. [H3C-voice-dial-anum2005]delay receive-dial 500 6.1.8 debugging voice vcc Syntax debugging voice vcc { all | cm | error | ipp | proc | rcv | timer | vpp | line line-number } undo debugging voice vcc { all | cm | error | ipp | proc | radius | rcv | timer | vpp | line line-number } View User view Parameter all: Enables all debugging of the VCC module. 6-6 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands cm: Enables the debugging of the messages between the CM module and the VCC module. error: Enables the error debugging of the VCC module. ipp: Enables the debugging of the messages between the IPP module and the VCC module. proc: Enables the debugging of the messages between the system process and the VCC module. radius: Enables the debugging of the messages between the RADIUS Client and the VCC module. rcv: Enables the debugging of the messages between the RCV module and the VCC module. timer: Enables the debugging information of the messages between the timer module and the VCC module. vpp: Enables the debugging information of the messages between the VPP module and the VCC module. line: Enables the debugging of the specified subscriber line on the router. line-number: Subscriber line number, the value range of it is decided by the type and quantity of the voice cards actually put into operation. Description Use the debugging voice vcc command to enable the debugging at various levels of the VCC module. This command is used to enable the debugging of the VCC module, and to specify the levels and types of the debugging. Note: When specifying the levels and types of the debugging, pay attention to the following points: z No character other than figures can appear in the specified line number, If any character other than figures appears in the entered line-number, this enter is invalid. z The specified line number cannot be greater than the number of the actual subscriber lines. If the specified line-number is greater than the number of the actual subscriber lines, this enter is invalid. Related command: reset voice vcc, display voice vcc. 6-7 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Example # Debug the messages sent to the VPP module from the VCC module. debugging voice vcc vpp 6.1.9 display aaa unsent-h323-call-record Syntax display aaa unsent-h323-call-record View Any view Parameter None Description Use the display aaa unsent-h323-call-record command to display the CDRs not sent successfully. If the RADIUS Client (namely, the router) is configured to process the Accounting Request/Response message in the start-ack mode, a call cannot be established on the VoIP side if the RADIUS Server fails to return an Accounting Response message within the stipulated time. The router will save the call failure and originate another Accounting Request message to the RADIUS Server after some time to attempt to establish a connection. This command is used to display the history records of calls that fail to be established on the VoIP side because the RADIUS server does not respond to the Accounting-Start message. The output information can help you to determine which calls fail and can also help you to analyze, locate, and remove the fault. Example # Display history records of calls that fail to be established on the VoIP side because the RADIUS server does not respond to the Accounting-Start message. [H3C] display aaa unsent-h323-call-record Index = 1 Acct_session_Id = 10 CallOrigin = Answer CallType = Telephony Callernumber = 1000 Callednumber = 1001 CallDuration = 00:00:03 TransmitPackets = 1000 TransmitBytes = 32000 6-8 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands ReceivePackets = 1100 ReceiveBytes = 35200 Table 6-1 Description on fields of the display aaa unsent-h323-call-record command Field Description Index Index of a history record of call failure Acct_session_Id ID of a RADIUS session, negotiated between RADIUS Server and RADIUS Client CallOrigin Call direction, which indicates whether the local end is the calling end or called end CallType Call type, including telephony and fax Callernumber Calling number Callednumber Called number CallDuration Call duration TransmitPackets Number of packets sent by the local end TransmitBytes Number of bytes sent by the local end ReceivePackets Number of packets received by the local end ReceiveBytes Number of bytes received by the local end 6.1.10 display voice aaa-client configuration Syntax display voice aaa-client configuration View Any view Parameter None Description Use the display voice aaa-client configuration command to view information about voice AAA. Example # Display information about voice AAA. [H3C] display voice aaa-client configuration AAA configuration : 6-9 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands accounting\authentication\authorization: accounting = off authentication-did = off accounting-method = start-stop dial-control-mib-info: buffer = 50 duration = 15 6.1.11 display voice call-history-record Syntax display voice call-history-record { callednumber called-number | callingnumber calling-number | cardnumber card-number | remote-ip-addr a.b.c.d | last last-number | line line-number } [ brief ] View Any view Parameter callednumber: Displays the call history records according to the called number filtering. called-number: The called E.164 number, being a character string within 31 characters. callingnumber: Displays the call history records according to the calling number filtering. calling-number: The calling E.164 number, being a character string within 31 characters. cardnumber: Displays the call history records according to the prepaid card number. card-number: The prepaid card number, being a character string within 31 characters. remote-ip-addr: Displays the call history records according to the IP address of the called user. a.b.c.d: The IP address of the called user, in the dotted decimal format. last: Displays the call history records according to the specified number of the latest conversation records. last-number: The number of the latest conversation records, in the value range from 0 to 500. line: Displays the call history records according to the voice subscriber-line receiving calls on the router. line-number: The number of the voice subscriber-line receiving calls on the router, and its value range is decided according to the type and quantity of voice cards actually put into operation on the router. 6-10 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands brief: Uses a brief format to display call history records. Description Use the display voice call-history-record command to view the information about call history records. If such parameters as callednumber, callingnumber, cardnumber, remote-ip-addr, last, line are not entered, all the saved call history records should be output by default. If it is specified that call history records are displayed according to the number of the latest conversation records (in “last” mode), the default number of records are 10. The display format of call history records can be customized by setting the display and filtering rule. The possible information display formats are as follows: 1) If the parameter brief is specified, the records will be displayed in a brief format. 2) If the parameter brief is not specified, the records will be displayed in a detailed format. If it is required to search according to the specified last-number, the records will be displayed according to the number of records actually retrieved. If no record is found or the record found is null, the prompting information as follows will appear: Not found any record. Note: When setting the display and filtering rule, pay attention to the following points: z No unacceptable character is allowed to appear in the calling number or called number. If any character other than figures and “*”, “#”, “T”, “.” appears in the entered called-number or calling-number, the enter is invalid. z No character other than figures is allowed to appear in the specified number of the latest conversation records times and voice subscriber-line number. If any character except figures appears in the entered last-number/line-number, the enter is invalid. z The digits of the specified called number and calling number cannot be over 31. If the entered called-number or calling-number has over 31 characters, the enter is invalid. z The specified number of the latest conversation records cannot be over 500. If the specified last-number is greater than 500, the enter is invalid. z The specified voice subscriber-line number cannot be greater than the number of available subscriber lines. If the specified line-number is greater than the number of available subscriber lines, the enter is invalid. The RADIUS Server may respond for the following reasons: z Normal release z Card number not exist 6-11 Command Manual – Voice Comware V3 z Invalid password z This accounts is using z No enough balance z The accounts is expired z Credit limit z User reject z Service invalid z Called limit Chapter 6 Voice RADIUS Configuration Commands Example # Display the call history records according to the number of 10 of the latest conversation records. [H3C] display voice call-history-record last 10 ! CallRecord[30]: CallerNum = 4000 CalledNum = 2000 EncodeType = 711u PeerAddress = 127.0.0.1 DisconnectCause = 0 DisconnectText = Normal release TalkingTimes = 00h 00m 00s VoiceTimes = 00h 00m 00s FaxTimes = 00h 00m 00s ImgPages = 0 CallDirection = 2 SetupTime(voip) = Mar 11, 2006 02:20:27 ConnectTime(voip) = None DisconectTime(voip) = Mar 11, 2006 02:20:29 Transmit (voip) = 0 Received(voip) = SetupTime(pstn) = None ConnectTime(pstn) = None DisconectTime(pstn) = None Transmit (pstn) = 0 Received(pstn) = 0 0 ! 6-12 (package) : 45456 (byte) (package) : 45456 (byte) (package) : 45456 (byte) (package) : 45456 (byte) Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Table 6-2 Description on the fields of the display voice call-history-record command Field Description CallRecord[30] Number of call record CallerNum Caller’s number CalledNum Called number EncodeType Encoding type PeerAddress Peer address DisconnectCause Code of disconnect cause DisconnectText Text of disconnect cause TalkingTimes Talking duration VoiceTimes Voice talking duration FaxTimes Fax talking duration ImgPages Facsimile pages CallDirection Calling direction SetupTime(voip) Call setup time on network side ConnectTime(voip) Call connecting time on network side DisconectTime(voip) Call disconnecting time on network side Transmit (voip) Number of packets transported on network side (packet/byte) Received(voip) Number of packets received on network side (packet/byte) SetupTime(pstn) Call setup time on PSTN side ConnectTime(pstn) Call connecting time on PSTN side DisconectTime(pstn) Call disconnecting time on PSTN side Transmit (pstn) Number of packets transported on PSTN side (packet/byte) Received(pstn) Number of packets (packet/byte) received on PSTN side 6.1.12 display voice vcc Syntax display voice vcc { channel [ line-number ] } | { statistic { all | error | ipp | proc | rcv | timer | vpp } } 6-13 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands View Any view Parameter channel: Displays the status information of the call channel on the specified subscriber line. line-number: Subscriber line number, the value range of it is decided by the type and quantity of the voice cards actually put into operation. statistic: Displays the statistics of the VCC module. all: Displays all the statistics of the VCC module. error: Displays the error statistics of the VCC module. ipp: Displays the statistics of the interaction messages between the VCC module and the IPP module. proc: Displays the statistics of the interaction messages between the VCC module and the system process. rcv: Displays the statistics of the interaction messages between the VCC module and the RCV module. timer: Displays the statistics of the interaction messages between the VCC module and the timer. vpp: Displays the statistics of the interaction messages between the VCC module and the VPP module. Description Use the display voice vcc command to view the information about the call channel status and call statistics. If it is required to display the status information of the call channel on the subscriber line, but the parameter line-number is not entered, the status information of all voice subscriber-lines will be displayed by default. Related command: reset voice vcc, debugging voice vcc. Note: When setting the filtering rule, pay attention to the following points: The specified voice subscriber-line number cannot be greater than the number of the actual subscriber lines. If the specified line-number is greater than the number of the actually available subscriber lines, this enter is invalid. 6-14 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Example # Display the call status information of the 2VI board voice subscriber-line 1. [H3C] display voice vcc channel 1 ! GENERIC: Index = 12 CallerNum = 100 CalledNum = 200 EncodeType = 729 PeerAddress = 127.0.0.1 DisconnectCause = 0 DisconnectText = Normal release CallDuration = 00h 00m 17s VoiceCallDuration = 00h 00m 17s FaxCallDuration = 00h 00m 00s ImgPages = 0 CallOrigin = 1 ! VoIP: SetupTime(voip) = Mar 29, 2006 15:37:21 ConnectTime(voip) = Mar 29, 2006 15:37:24 TransmitPackets(voip) = 531 TransmitBytes(voip) = 17126 ReceivePackets(voip) = 550 ReceiveBytes(voip) = 23100 ! PSTN: SetupTime(pstn) = Mar 29, 2006 15:37:06 TransmitPackets(pstn) = 550 TransmitBytes(pstn) = 16500 ReceivePackets(pstn) = 913 ReceiveBytes(pstn) = 18130 Table 6-3 Description on call channel status Field Description Index Index of current call in channel CallerNum Caller number of current call in channel CalledNum Called number of current call in channel EncodeType Compression/Decompression type of current call in channel 6-15 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Field Description PeerAddress IP address of Peer GW of current call in channel CallDuration Call duration of current call in channel VoiceTxDuration Total voice duration of current call in channel at PSTN side FaxTxDuration Total fax duration of current call in channel at PSTN side ImgPages Total pages of sending/receiving fax CallOrigin Direction of current call in channel, it means local end is caller end or called end corresponding with 1 or 2 respectively SetupTime(voip/pstn) Time of current call is setup(VoIP/PSTN) ConnectTime(voip/pstn) Connect start time channel(VoIP/PSTN) TransmitPackets Number of packets sent at local end of current call in channel TransmitBytes Number of bytes sent at local end of current call in channel ReceivePackets Number of packets received at local end of current call in channel ReceiveBytes Number of bytes received at local end of current call in channel of current call in 6.1.13 gw-access-number Syntax gw-access-number access-number undo gw-access-number [ access-number ] View Voice dial program view Parameter access-number: The specified access service number (such as 169, 17900, etc.). The value is a character string within 31 characters, and the acceptable symbols include the figures from 0 to 9, the wildcard “.” and the character “T”, and other English characters. The “.” wildcard indicates a character. The character “T" is control character, it indicating that the dial string is a string whose length is variable. At most 100 access service numbers can be configured. 6-16 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Description Use the gw-access-number command to configure the access service number or enter the view of the access service number. Use the undo gw-access-number command to cancel a single access service number or all access service numbers that have been set. If no access service number is specified in the undo gw-access-number command, all existing access service numbers are deleted. If an access service number is specified, only this number is deleted. When you attempt to delete all access service numbers that have been set, the system gives the following operation warning: Delete all of the access number? (n/y) Select “y” to confirm the delete or “n” to cancel the delete. Note: When setting an access service number, pay attention to the following points: z No unacceptable character is allowed to appear in the access service numbers, if any of these characters (such as English characters, etc.) appears in the entered access-number, this enter is invalid. z The digits of an access service number cannot be more than 31, if those of the entered access-number are over 31, this enter is invalid. z At most 100 access service numbers are allowed to be set in the system, if 100 pieces of access-number have been set, no more access service number can be added. To continue with adding settings, please delete some records. Related command: process-config, card-digit, password-digit, redialtimes. Example # Add the access service number 18901 and enter the voice access-number view. [H3C-voice-dial] gw-access-number 18901 [H3C-voice-dial-anum18901] # Delete all the access service numbers that have been set in the system. [H3C-voice-dial] undo gw-access-number 6.1.14 password-digit Syntax password-digit password-digit 6-17 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands View Voice access-number view Parameter password-digit: The password digit, in the range from 1 to 16.By default, the user password is in 6 digits. Description Use the password-digit command to configure the user password digits of a certain access service number in the card number process. This command is used to configure the password digit of two-stage dialing card number process (use the card number/password to authenticate a user’s ID) users. If the dialing process of the access service number to be configured is not specified as the card number process through the process-config command, this command cannot take effect. Only after the dialing process is set as the card number process, can the access service number be retrieved, and can this command function properly. Related command: gw-access-number, process-config, card-digit. Example # Set the password in 4 digits for the access service number 18901. [H3C-voice-dial-anum18901] password-digit 4 6.1.15 process-config Syntax process-config { callernumber | cardnumber | voice-caller } View Voice access-number view Parameter callernumber: Caller number process. After a user dials the access service number, the system continues to send dialing tones to the user, so that he can enter the called number. Under this process, the user’s ID is authenticated by identifying the caller number. cardnumber: Card number process. After a user dials the access service number, the system continues to send prompting tones to the user, so that he can enter the prepaid card number and password. Under this process, the user’s ID is authenticated by identifying the prepaid card number/password. 6-18 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands voice-caller: Voice caller number process. Configured with it, the system plays voice messages to users after they dial the access number, asking them to select language and input the called numbers. In this process, users are authenticated using calling number identification (CNI). When the user changes the current dial process (that is, between the calling number dial process and the card number dial process), the system restores the involved parameters (for example, card number/PIN, and redial times in the card number process) to the defaults automatically. Description Use the process-config command to configure the dialing process of a certain access service number. By default, all access service numbers use the card number process. Each access service number has a specific dialing process. For a certain access service number, all the users to whom it belongs must implement call establishment in the same process. With Comware, three dial processes are available: calling number process, card number process, and voice caller number process. z Caller number process: It is actually the caller number authentication process, which is the authentication, authorization and accounting process carried out according to the caller number of the user. The caller number process does not require further parameter setting in the process. z Voice caller number process: uses calling numbers for authentication. After a user dials the access number, the voice gateway plays voice messages to prompt the user to select prompt language and dial the called number. This is different from the caller number process where the voice gateway displays only dial tone (long tone) after a user dials the access number. z Card number process: As its name implies, the user should dial its own card number and password to complete the authentication process after dialing an access service number, and he cannot dial the called number to set up a call until it is authenticated. The parameters in a card number process can be set through the such commands as card-digit, password-digit and redialtimes. Related command: gw-access-number, card-digit, password-digit, redialtimes. Example # Specify the user access process of the access service number 18901 as the card number process. [H3C-voice-dial-anum18901] process-config cardnumber 6-19 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands 6.1.16 redialtimes Syntax redialtimes redialtimes-number View Voice access-number view Parameter redialtimes-number: The number of times of dialing in each dialing phase in a card number process, in the value range from 1 to 10. By default, the number can be dialed three times in each dialing phase. Description Use the redialtimes command to configure the number of times of dialing in each dialing phase for a certain access service number. When the card number procedure or the caller number procedure with IVR is adopted, the user, after getting through to the access number, needs to follow voice prompts to operate. To avoid mistake in a dial segment from causing the failure of the entire dial process, you may specify the maximum number of dial attempts that a user can make to provide the correct number in each dial segment. For the caller number procedure without IVR, skip this configuration. Related command: gw-access-number, process-config, card-digit, password-digit. Note: z For the caller number procedure, the number of dial attempts allowed in each segment also makes sense. The setting can limit the number of attempts that the user can make to dial the called number after getting through to the access number. z For the card number procedure and the caller number procedure with IVR, the setting is effective for each dial segment. For example, for the card number procedure, the numbers of attempts that the user can make to input card number, password, and called number are the same. z The redialtimes-number argument specifies the number of dial attempts rather than redials. For example, to allow a user to redial three times in each dial segment, set the redialtimes-number argument to 4. 6-20 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Example # Configure the number of redialing times of the card number/password of the access service number 18901 to be 4 (the number can be dialed for 5 times). [H3C-voice-dial-anum18901] redialtimes 5 6.1.17 reset voice vcc Syntax reset voice vcc { all | call-record | statistics } View User view Parameter all: Clears all information. call-record: Clears the call record information. statistics: Clears the statistics. Description Use the reset voice vcc command to reset the information related to VCC. All or part of the information related to VCC module which is stored in the system can be cleared according to the information categories. Related command: display voice vcc, debugging voice vcc. Example # Clear the generated call record information. reset voice vcc call-record 6.1.18 selectlanguage Syntax selectlanguage { disable | enable } View Voice access-number view Parameter disable: Disables language selection. enable: Enables language selection. 6-21 Command Manual – Voice Comware V3 Chapter 6 Voice RADIUS Configuration Commands Description Use the selectlanguage command to enable or disable the voice gateway to play the prompt language selection message in the voice caller number process. This command only applies to the voice caller number process. With language selection enabled, the voice gateway plays a message asking the user to select a prompt language and then to input the called number. If language selection is disabled, the called number input prompt is played in Chinese after authentication. By default, language selection is disabled. Example # Disable prompt language selection. [H3C-voice-dial-anum163] selectlanguage disable # Enable prompt language selection. [H3C-voice-dial-anum163] selectlanguage enable 6-22 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands Chapter 7 GK Client Configuration Commands 7.1 GK Client Configuration commands 7.1.1 area-id (in Voice GK Client View) Syntax area-id string undo area-id [string ] View Voice GK client view Parameter string: Indicates the area ID. The value range is the integer between 0 and 9, which can be separated by “,” or "#". Description Use the area-id command to configure the H.323 gateway area ID. Use the undo-area-id command to cancel all area IDs. Use the undo-area-id string command to cancel the specified area ID. By default, no H.323 gateway area ID is configured. The area-id is mainly used to facilitate the identifying of gateway type by the GK Server. The gateway and the GK Server reach an agreement on the gateway type in advance, for example, consider that the area-id 1# represents the voice gateway and that the area-id 2# represents the video gateway, etc. When the gateway communicates normally with the GK Server, the GK will judge the gateway type according to the area-id information sent by the gateway. To set the H.323 area ID in the Voice GK client voice is mainly to facilitate the GK Server to identify the types of GK client. An agreement on the related types is reached beforehand between the GK client and the GK Server, e.g. area ID 1# represents a voice GK client, prefix 2# represents a video GK client, etc. When a normal communication is going on between the GK client and the GK Server, the GK judges the client type according to the area ID information received. Up to 30 area-ides can be configured in voice GK client view. Related command: match-template, entity. 7-1 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands Example # Configure the GK client with a area ID 6#. [H3C-voice-gk] area-id 6# 7.1.2 debugging voice ras Syntax debugging voice ras event View User view Parameter event: Output interacted RAS message record. Description Use the debugging voice ras command to enable the debugging information output switch of RAS messages interacted between GK Client and GK Server. Example # Enable the debugging information output switch of RAS messages interacted between GK Client and GK Server. debugging voice ras event 7.1.3 display voice gateway Syntax display voice gateway View Any view Parameter None Description Use the display voice gateway command to view the gateway registration state information to GK Server. This command is used to display the gateway registration information to GK Server and gateway alias name list etc. 7-2 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands Example # Display the gateway registration state information to GK Server. [H3C] display voice gateway GW_Statue = Registed GK_ID 8040gk.Comware.com = Current GW information : H323-ID 3681gw E164 34601000 Current GK Information: H323-ID 3681gw Table 7-1 Description of gateway registration state information to GK Server Field Description GW_Statue Status of the current GW terminal GK_ID ID of the currently registered GK Current GW information : H323-ID H323 ID of the current GW Current GW information : E164 E164 number of the current GW Current GK Information: H323-ID Local H323ID fed back by GK Current GK Information: E164 Local E164 number fed back by GK NONE None 7.1.4 gk-client Syntax gk-client View Voice view Parameter None Description Use the gk-client command to enter Voice GK client view, and configure the GK parameters for voice. Use the quit command to exit this view. Related command: area-id, gk-2nd-id, gk-id, gw-address, gw-id, ras-on. 7-3 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands Example # Enter gk-client view. [H3C-voice] gk-client [H3C-voice-gk] 7.1.5 gk-2nd-id Syntax gk-2nd-id gk-name gk-addr gk-ipaddress [ ras-port ] undo gk-2nd-id View Voice GK client view Parameter Refer to the parameters of the gk-id command. Description Use the gk-2nd-id command to configure the name and IP address of the backup GK Server corresponding to the gateway. Use the undo gk-2nd-id command to cancel the name and IP address of the backup GK Server corresponding to the gateway. When the communication between the GK Client and the master GK Server is abnormal (e.g. timeout) or the master GK Server fails, the H3C router can also send registration requests to the backup GK Server. Use the gk-2nd-id command to configure the IP address, name and port of the backup GK Server. Note: The gk-id command must be used first to configure the name and address of the master GK Server, before those of the backup GK Server can be configured. Related command: gk-id. Example # Configure the backup GK Server as follows: The IP address is 1.1.1.2, the name is "gk-backup", and use the default port. [H3C-voice-gk] gk-id gk-center gk-addr 1.1.1.1 7-4 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands [H3C-voice-gk] gk-2nd-id gk-backup gk-addr 1.1.1.2 7.1.6 gk-id Syntax gk-id gk-name gk-addr gk-ipaddress [ ras-port ] undo gk-id View Voice GK client view Parameter gk-name: GK Server name, a string of 1 to 128 case sensitive characters. gk-ipaddress: IP address of the GK Server. ras-port: RAS communication port of GK Server, with the value range being the integer between 1 and 65535. By default ras-port is 1719. Description Use the gk-id command to configure the GK Server name and IP address corresponding to the gateway. Use the undo gk-id command to cancel the GK Server name and IP address corresponding to the gateway. Use the command gk-id to configure such information as the IP address, name, and port of GK Server to facilitate the research for the right GK Server equipment by the GK Client according to this information, so as to implement the register task of gateway in the GK Server. Related command: area-id, gw-id, gk-2nd-id, gw-address, ras-on. Example # Configure the IP address of the GK Server as 1.1.1.1, name as gk-center, port as the default. [H3C-voice-gk] gk-id gk-center gk-addr 1.1.1.1 7.1.7 gk-security call Syntax gk-security call enable undo gk-security call enable 7-5 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands View Voice GK client view Parameter None Description Use the gk-security call enable command to enable security calls on the GK Client (router). Use the undo gk-security call enable command to disable security calls on the GK Client (router). By default, enable security calls. To let the router acting as a GK client pass calling tokens of a GK Server, you need to configure GK security calls. After a call is originated, the calling gateway obtains the calling token from the calling GK server and transparently transports it to the called gateway that will then pass the token to the called GK server. In some voice network environments, the called GK Servers cannot process calling tokens; in this case you must disable the GK Client security calls. Example # Disable the security calls on the GK Client (router). [H3C-voice-gk] undo gk-security call enable 7.1.8 gk-security register-pwd Syntax gk-security register-pwd { cipher | simple } password undo gk-security register-pwd View Voice GK client view Parameter cipher: Uses encrypted password when echo is adopted. simple: Do not use encrypted password when echo is adopted. password: Password that has been set, 1 to 16 printable characters, except space. Description Use the gk security register-pwd command to set the GK register password. 7-6 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands Use the gk-security register-pwd command to remove the GK register password. By default, the GK Client (router) has no password. After the GK Client (router) is configured with the register password, the password is carried during the whole register process. Example # Configure a GK register password to comware in cipher-text mode. [H3C-voice-gk] gk-security register-pwd cipher comware 7.1.9 gw-address Syntax gw-address { ip-address | interface interface-name } undo gw-address [ interface ] View Voice GK client view Parameter ip-address: Source address of the gateway. interface interface-name: Name of the interface to be bound. Description Use the gw-address command to bind the source address of the voice gateway to a permanent IP address or interface. Use the undo gw-address [ interface ] command to remove the binding. By default, the source IP address of the voice gateway is not bound. Caution: This command takes effect immediately after it is configured, disregarding whether the GK client has been enabled (using the ras-on command) or not. Related command: area-id, gk-2nd-id, gk-id, gw-address, ras-on. Example # Bind the source IP address of the voice gateway to 1.1.1.1. [H3C-voice-gk] gw-address 1.1.1.1 7-7 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands # Bind the source address of the voice gateway to Ethernet0/0. [H3C-voice-gk] gw-address interface Ethernet0/0 7.1.10 gw-id Syntax gw-id namestring undo gw-id View Voice GK client view Parameter namestring: Gateway alias (gateway ID), a string of 1 to 128 case sensitive characters. Description Use the gw-id command to configure the gateway alias. Use the undo gw-id command to cancel the alias of the specified gateway. By default, the gateway alias is empty, i.e., the interface alias is not configured. Use the command gw-id to configure the gateway alias, which is used for the gateway to register and identify the voice gateway. Each gateway has only one alias, and the new alias will cover the old one. Related command: area-id, gk-2nd-id, gk-id, gw-address, ras-on. Example # Configure the gateway alias as beijing-gw. [H3C-voice-gk] gw-id beijing-gw 7.1.11 ras-on Syntax ras-on undo ras-on View Voice GK client view Parameter None 7-8 Command Manual – Voice Comware V3 Chapter 7 GK Client Configuration Commands Description Use the ras-on command to enable the GK Client function. Use the undo ras-on command to disable the GK Client function. By default, the GK Client function is disabled. Only after activating the GK Client function can the normal communication be maintained between the router voice gateway and the GK server, or it is unable to establish the connection between them. Example # Activate the GK Client function. [H3C-voice-gk] ras-on 7-9 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands Chapter 8 SIP Client Commands 8.1 SIP Client Commands 8.1.1 address sip Syntax address sip { proxy | { ip ip-address port port-number } } undo address sip proxy | ip View Voice entity view Parameter proxy: Adopts SIP proxy server to complete SIP interaction. ip-address: IPv4 address to which SIP packets of a call is destined. port port-number: Port number in the range 1 to 65535. Description Use the address sip command to set routing policy for reaching the remote voice gateway (VG) to SIP. Use the undo address command to restore the default. By default, no routing policy is configured for reaching the remote VG. If address sip ip ip-address port port-number is configured, SIP packets in a call are routed directly to the address specified in the command. If address sip proxy is configured, the SIP interaction process involves a SIP proxy server. Configure the command appropriate to the call routing policy applies to the network. Related command: address. Note: After configuring this command, the area-id command and the fast-connect command, if configured, do not take effect any more. 8-1 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands Example # Configure the current voice entity to complete SIP interaction through a SIP proxy server. [H3C-voice-dial-entity1] address sip proxy 8.1.2 debugging voice sip Syntax debugging voice sip { all | calls | error | message { all | call | register } | register | warning } undo debugging voice sip { all | calls | error | message { all | call | register } | register | warning } View User view Parameter all: Enables all SIP debugging. calls: Enables SIP call debugging. error: Enables SIP error debugging. message: Enables debugging for the detailed information received and transmitted by SIP. register: Enables SIP register debugging. warning: Enables SIP warning debugging. Description Use the debugging voice sip command to enable SIP debugging. Use the undo debugging voice sip command to disable SIP debugging. Example # Enable all SM SIP debugging. debugging voice sip all 8.1.3 display voice sip call-statistics Syntax display voice sip call-statistics 8-2 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands View Any view Parameter None Description Use the display voice sip call-statistics command to view all statistic information about the SIP client. Example # Display all statistic information about the SIP client. [H3C] display voice sip call-statistics SIPRegisters : 0 SIPInvites : SIPAcks : 0 0 SIPPracks : SIPByes : 0 0 SIPCancels : SIPInfos : 0 0 SIPOKRegisters : SIPOKInvites : SIPOKAcks : 0 0 SIPOKPracks : SIPOKByes : 0 0 0 SIPOKCancels : 0 SIPResp3MultipleChoice : 0 SIPResp3MovedPermanently : 0 SIPResp3MovedTemporarily : 0 SIPResp3UseProxy : SIPResp3Other : 0 0 SIPResp4BadRequest : 0 SIPResp4Unauthorized : SIPResp4Forbidden : SIPResp4NotFound : 0 0 0 SIPResp4MethodNotAllowed : SIPResp4NotAcceptable : 0 0 SIPResp4ProxyAuthRequired : SIPResp4ReqTimeout : 0 0 SIPResp4ReqEntityTooLarge : SIPResp4ReqURITooLarge : 0 0 SIPResp4UnsupportedMediaType : 0 8-3 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands SIPResp4UnsupportedURIScheme : SIPResp4BadExtension : 0 SIPResp4ExtensionRequired : SIPResp4AddrIncomplete : SIPResp4BusyHere : 0 0 0 SIPResp4RequestTerminated : SIPResp4Other : 0 0 0 SIPResp5InternalError : 0 SIPResp5NotImplemented : SIPResp5BadGateway : 0 0 SIPResp5ServiceUnavailable : SIPResp5GatewayTimeout : SIPResp5BadSipVersion : SIPResp5MessageTooLarge : SIPResp5Other : SIPResp6xx : 0 0 0 0 0 0 See the following table for field description. Table 8-1 Description of the fields of the display voice sip call-statistics command Field Description SIPRegisters REGISTER requests sent and received by the SIP gateway SIPInvites INVITE requests sent and received by the SIP gateway SIPAcks ACKs sent and received by the SIP gateway SIPPracks PACK requests sent and received by the SIP gateway SIPByes BYE requests sent and received by the SIP gateway SIPCancels CANCEL requests sent and received by the SIP gateway SIPInfos INFO requests sent and received by the SIP gateway SIPOKRegisters Successful REGISTER requests sent and received by the SIP gateway SIPOKInvites Successful INVITE requests sent and received by the SIP gateway SIPOKAcks Successful ACK requests sent and received by the SIP gateway SIPOKPracks Successful PACK requests sent and received by the SIP gateway 8-4 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands Field Description SIPOKByes Successful BYE requests sent and received by the SIP gateway SIPOKCancels Successful CANCEL requests sent and received by the SIP gateway SIPResp3MultipleChoice 300 (Multiple Choices) responses received by the SIP gateway SIPResp3MovedPermanently 301 (Moved Permanently) responses received by the SIP gateway SIPResp3MovedTemporarily 302 (Moved Temporarily) responses received by the SIP gateway SIPResp3UseProxy 305 (Use Proxy) responses received by the SIP gateway SIPResp3Other Other 3xx responses received by the SIP gateway SIPResp4BadRequest 400 (Bad Request) responses received by the SIP gateway SIPResp4Unauthorized 401 (Unauthorized) responses received by the SIP gateway SIPResp4Forbidden 403 (Forbidden) responses received by the SIP gateway SIPResp4NotFound 404 (Not Found) responses received by the SIP gateway SIPResp4MethodNotAllowed 405 (Method Not Allowed) responses received by the SIP gateway SIPResp4NotAcceptable 406 (Not Acceptable) responses received by the SIP gateway SIPResp4ProxyAuthRequire d 407 (Proxy Authentication Required) responses received by the SIP gateway SIPResp4ReqTimeout 408 (Request Timeout) responses received by the SIP gateway SIPResp4ReqEntityTooLarge 413 (Request Entity Too Large) responses received by the SIP gateway SIPResp4ReqURITooLarge 414 (Request-URI Too Long) responses received by the SIP gateway SIPResp4UnsupportedMedia Type 415 (Unsupported Media received by the SIP gateway Type) responses SIPResp4UnsupportedURISc heme 416 (Unsupported URI Scheme) received by the SIP gateway responses SIPResp4BadExtension 420 (Bad Extension) responses received by the SIP gateway 8-5 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands Field Description SIPResp4ExtensionRequired 421 (Extension Required) responses received by the SIP gateway SIPResp4AddrIncomplete 484 (Address Incomplete) responses received by the SIP gateway SIPResp4BusyHere 486 (Busy Here) responses received by the SIP gateway SIPResp4RequestTerminate d 487 (Request Terminated) responses received by the SIP gateway SIPResp4Other Other 4xx responses received by the SIP gateway SIPResp5InternalError 500 (Server Internal Error) responses received by the SIP gateway SIPResp5NotImplemented 501 (Not Implemented) responses received by the SIP gateway SIPResp5BadGateway 502 (Bad Gateway) responses received by the SIP gateway SIPResp5ServiceUnavailable 503 (Service Unavailable) responses received by the SIP gateway SIPResp5GatewayTimeout 504 (Server Time-Out) responses received by the SIP gateway SIPResp5BadSipVersion 505 (Version Not Supported) responses received by the SIP gateway SIPResp5MessageTooLarge 513 (Message Too Large) responses received by the SIP gateway SIPResp5Other Other 5xx responses received by the SIP gateway SIPResp6xx 6xx responses received by the SIP gateway 8.1.4 mode (SIP client) Syntax mode {gateway { all | single } | phone number number } undo mode View SIP Client view Parameter gateway: Sets the registration mode of the gateway to gateway. all: All telephone numbers are carried in one register message. 8-6 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands single: One telephone number is carried in one register message. phone: Sets the registration mode of the gateway to phone. number number: The number to be registered in phone mode. Description Use the mode command to set the registration mode of the gateway. Use the undo mode command to restore the default registration mode, or the gateway mode. In gateway mode, the router can register multiple numbers while in phone mode it can register only the one specified by the number argument. Example # Set the gateway registration mode to phone and the number to be registered to 1688. [H3C-voice-sip] mode phone number 1688 8.1.5 register-enable Syntax register-enable { on | off } undo register-enable View SIP client view Parameter on: Enables the SIP registration function. off: Disables the SIP registration function. Description Use the register-enable command to enable or disable the SIP registration function. By default, the SIP registration function is not enabled. You can disable the SIP registration function by executing either undo register-enable or register-enable off. Example # Enable the SIP registration function. [H3C-voice-sip] register-enable on 8-7 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands 8.1.6 reset voice sip Syntax reset voice sip Parameter None View User view Description Use the reset voice sip command to reset all the statistic information about the SIP client. Example # Reset all the statistic information about the SIP client. reset voice sip 8.1.7 sip Syntax sip View Voice view Parameter None Description Use the sip command to access SIP client view. Before you can configure a SIP client, you should first access the appropriate SIP client view. Example # Access SIP client view. [H3C-voice] sip [H3C-voice-sip] 8-8 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands 8.1.8 sip-call forwarding Syntax sip-call forwarding undo sip-call forwarding View SIP Client view Parameter forwarding: Enables call forwarding. waiting: Enables call waiting. transfer: Enables call transfer. Description Use the sip-call forwarding command to enable call forwarding. Use the undo sip-call forwarding command to disable call forwarding. By default, SIP call forwarding is disabled. Example # Enable SIP call forwarding. [H3C-voice-sip] sip-call forwarding 8.1.9 sip-comp Syntax sip-comp { callee | from | server } undo sip-comp { callee | from | server } View SIP client view Parameter callee: Extracts the called number from the To field. from: Configures the device to use the IP address in the To field as the IP address in the From field when sending a SIP request. By default, the From field indicates the calling address and the To field indicates the called address. server: Allows the Server field to be included in requests. According to RFC 3261, the Server field is an optional field in responses. 8-9 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands Description Use the sip-comp command to configure a SIP compatibility option. Use the undo sip-comp command to restore the default setting of a SIP compatibility option. Note: The sip-comp from command can be used in the case that voice gateway has passed sip proxy call authentication and the IP addresses of From and To field are the same. Do not configure this command in other cases. Example # Configure the device to use the IP address in the To field as the IP address in the From field when sending a SIP request. [H3C-voice-sip] sip-comp from 8.1.10 sip-comp agent Syntax sip-comp agent [agent-info ] undo sip-comp agent View SIP Client view Parameter agent: Includes the user-agent field in SIP request. agent-info: Character string, in the range of 1 to 31, indicating the user-agent field information, which defaults to H3C-Router. Description Use the sip-comp agent command to configure including user-agent field in SIP request. Use the undo sip-comp command to configure not including user-agent field in SIP request. By default, user-agent field is not included in SIP request. 8-10 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands Example # Configure the user-agent field to be included in SIP request, and the value is h3c. [H3C-voice-sip] sip-comp agent h3c # Configure the user-agent field to be included in SIP request, and uses the default value. [H3C-voice-sip] sip-comp agent # Configure the user-agent field not to be included in SIP request. [H3C-voice-sip] undo sip-comp agent 8.1.11 sip-domain Syntax sip-domain domain-name undo sip-domain View SIP client view Parameter domain-name: Domain name of the SIP server, in the range 1 to 31 characters. Description Use the sip-domain command to configure the domain name of a SIP server. Use the undo sip-domain command to delete the domain name of the SIP server. By default, a SIP server is identified by its IP address instead of domain name. Example # Configure a SIP server’s domain name hello.com. [H3C-voice-sip] sip-domain hello.com 8.1.12 sip-id Syntax sip-id id password { simple | cipher } password undo sip-id View SIP client view 8-11 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands Parameter id: SIP ID of router, which is a case sensitive character string in the range 0 to 9, a to z, A to Z, !, %, *, -, _, +, ‘, ., and @. @ cannot be used as the last character. ID ranges from 1 to 31 and defaults to Comware-GATEWAY. simple: Displays the configuration of SIP-ID password in plain text. cipher: Displays the configuration of SIP-ID password in ciphertext. password: SIP authentication password of the gateway, which defaults to Comware-SIP. Description Use the sip-id command to set ID of the local gateway and configure its SIP authentication password. Use the undo sip-id command to restore the default. By default, SIP ID of gateway is Comware-GATEWAY and authentication password is Comware-SIP. During the SIP message interaction, the local device may be required by the remote SIP device to provide SIP ID and password for authentication. You can use this command however only when the register-enable off command is configured. Example # Set SIP ID of the router to h3c, password to 1234, and display mode to ciphertext. [H3C-voice-sip] sip-id h3c password cipher 1234 8.1.13 sip-server Syntax sip-server { master | slaver } ip-address port [ port-number | default ] [ inbound | all ] undo sip-server View SIP client view Parameter master: Master proxy server. slaver: Slave proxy server. ip-address: IPv4 address. port port-number: Port number, which is in the range 1 to 65535 and defaults to 5060. inbound: Accepts only SIP call requests from the specified server. 8-12 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands all: Accepts all the SIP call requests. Description Use the sip-server command to configure address information of SIP proxy server and define call requests that can be accepted. Use the undo sip-server command to restore the default. By default, all the SIP call requests are accepted. You can use this command to configure master and slave SIP proxy servers. During an interaction, requests are sent to the slave proxy server if the master is not available. If the inbound keyword applies, only SIP call requests from the specified proxy server are acceptable; if the all keyword applies, all the SIP call requests are acceptable. You can use this command however only when the register-enable off command is configured. The address configured here should be consistent with that assigned to the specified SIP proxy server at the time of networking. Example # Set the address of master SIP proxy server to 169.54.5.10 and port number to 1120. [H3C-voice-sip] sip-server master 169.54.5.10 port 1120 all 8.1.14 source-ip Syntax source-ip ip-address undo source-ip View SIP client view Parameter ip-address: IPv4 address. Description Use the source-ip command to bind a source IP address to the gateway when it functions as a User Agent (UA). By default, no SIP source IP address is configured. Note that this source address should be the IP address of an existing interface on the gateway. 8-13 Command Manual – Voice Comware V3 Chapter 8 SIP Client Commands You can use this command however only when the register-enable off command is configured. Example # Set the local source address to 1.1.1.1. [H3C-voice-sip] source-ip 1.1.1.1 8.1.15 wildcard-register enable Syntax wildcard-register enable undo wildcard-register View SIP client view Parameter None Description Use the wildcard-register enable command to enable fuzzy (wildcard) telephone number registration. Use the undo wildcard-register command to disable fuzzy (wildcard) telephone number registration. By default, fuzzy telephone number registration is disabled. Sometimes, the match template configured for a POTS entity may use a number containing the wildcards of dot (.) and T instead of using a standard E.164 number. When the router sends a Register message, it retains dots and substitutes dots (.) for Ts. Note: You may use the function of fuzzy telephone number registration only when it is supported on both SIP client and SIP server. Example # Enable fuzzy telephone number registration. [H3C-voice-sip] wildcard-register enable 8-14