HP A-MSR Router Series Voice Command Reference

Abstract This document describes the commands and command syntax options available for the HP A Series products. This document is intended for network planners, field technical support and servicing engineers, and network administrators who work with HP A Series products.

Part number: 5998-2047 Software version: CMW520-R2207P02 Document version: 6PW100-20110810

Legal and notice information

Copyright 201 Hewlett-Packard Development Company, L.P. 1 No part of this documentation may be reproduced or transmitted in any form or by any means without prior written consent of Hewlett-Packard Development Company, L.P. The information contained herein is subject to change without notice. HEWLETT-PACKARD COMPANY MAKES NO WARRANTY OF ANY KIND WITH REGARD TO THIS MATERIAL, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. Hewlett-Packard shall not be liable for errors contained herein or for incidental or consequential damages in connection with the furnishing, performance, or use of this material. The only warranties for HP products and services are set forth in the express warranty statements accompanying such products and services. Nothing herein should be construed as constituting an additional warranty. HP shall not be liable for technical or editorial errors or omissions contained herein.

Contents
Voice entity configuration commands 1 call-history 1 compression 1 default entity compression 7 default entity payload-size 8 default entity vad-on 9 description (voice entity view) 10 dial-trap enable 10 dial-program 11 display voice call-info 11 display voice cmc 13 display voice default all 16 display voice entity 17 display voice ipp statistic 19 display voice iva statistic 21 display voice statistics call-active 22 display voice statistics call-history 25 display voice statistics entity 28 distinguish-localtalk 30 dscp media 30 entity 31 line 32 match-template 32 outband 35 payload-size 35 register-number 36 reset voice cmc statistic 37 reset voice ipp statistic 37 reset voice iva statistic 38 rtp payload-type nte 38 send-ring 39 shutdown (voice entity view) 40 vad-on 40 voice-setup 41 voip timer 42 vqa dscp 42 vqa dsp-monitor buffer-time 44 Voice subscriber line configuration commands 45 Analog voice subscriber line configuration commands 45 area 45 busytone-hookon timer 46 busytone-t-th 46 calling-name 47 cid display 48 cid receive 48 cid ring 49 cid send 50 cid type 50 cng-on 51
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cptone country-type 52 cptone tone-type 54 default 55 default subscriber-line 56 delay hold 56 delay rising 57 delay send-dtmf 58 delay send-wink 58 delay wink-hold 59 delay wink-rising 59 delay start-dial 60 description (voice subscriber line view) 61 disconnect lcfo 61 display voice subscriber-line 62 dtmf amplitude 65 dtmf sensitivity-level 65 dtmf time 66 dtmf threshold 67 echo-canceller 69 echo-canceller parameter 70 em-phy-parm 71 em-signal 71 em-passthrough 72 hookoff-mode 72 hookoff-mode delay bind 73 hookoff-time 74 impedance 74 nlp-on 75 open-trunk 76 plc-mode 77 receive gain 78 reset voice cmc statistic 78 reset voice ipp statistic 79 reset voice iva statistic 79 ring-detect debounce 80 ring-detect frequency 81 send-busytone 81 shutdown (voice subscriber line view) 82 silence-th-span 83 slic-gain 83 subscriber-line 84 timer dial-interval 84 timer disconnect-pulse 85 timer first-dial 85 timer hookflash-detect 86 timer hookoff-interval 87 timer ring-back 87 timer wait-digit 88 transmit gain 88 type 89 vi-card busy-tone-detect 90 vi-card cptone-custom 91 vi-card reboot 92 Digital voice subscriber line configuration commands 93 amd enable 93
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amd parameter 93 ani 94 ani-offset 95 answer enable 96 callmode 96 cas 97 clear-forward-ack enable 98 display voice subscriber-line 99 dl-bits 100 dtmf enable 102 dtmf threshold digital 102 enable snmp trap updown 103 final-callednum enable 104 force-metering enable 104 group-b enable 105 line 106 mode 106 pcm 108 posa called-length 108 pri-set 109 qsig-tunnel enable 110 re-answer enable 110 register-value 111 renew 113 reverse 114 seizure-ack enable 115 select-mode 115 sendring ringbusy enable 116 signal-value 117 special-character 118 subscriber-line 119 tdm-clock 119 timer dl 120 timer dtmf 121 timer hold 122 timer register-pulse persistence 123 timer register-complete group-b 124 timer ring 124 timeslot-set 125 trunk-direction 126 ts 127

Dial plan configuration commands 129 caller-group 129 caller-permit 129 description 131 dial-prefix 132 display voice subscriber-group 133 display voice number-substitute 134 dot-match 135 first-rule 136 match-template 136 max-call (voice dial program view) 138 max-call (voice entity view) 139 number-match 139
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number-priority 140 number-substitute 141 priority 141 private-line 142 rule 143 select-rule rule-order 147 select-rule search-stop 148 select-rule type-first 149 select-stop 150 send-number 150 subscriber-group 151 substitute (voice subscriber line view, voice entity view) 152 substitute (voice dial program view) 153 terminator 154

SIP configuration commands 155 address sip 155 call-fallback 156 crypto 156 display voice sip call-statistics 157 display voice sip connection 160 display voice enum-group 161 display voice sip dns-record 162 display voice sip reason-mapping 162 dns-type 165 display voice sip register-state 166 early-media enable 167 enum-group 168 keepalive 168 line-check enable 169 listen transport 170 media-protocol 171 outband sip 171 outbound-proxy 172 privacy 173 proxy 173 reason-mapping pstn 174 reason-mapping sip 176 register-enable 178 redundancy mode 179 registrar 179 remote-party-id 181 reset voice sip connection 181 reset voice sip dns-record 182 reset voice sip statistics 182 rule 183 sip 183 sip-comp 184 sip-comp agent 185 sip-comp server 186 sip-domain 186 source-bind 187 timer connection age 188 timer registration retry 188 timer registration expires 189
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timer registration divider 189 timer registration threshold 190 timer session-expires 191 transport 191 uri 192 url 193 user 194 wildcard-register enable 195

SIP local survival configuration commands 197 area-prefix 197 authentication 197 call-route 198 call-rule-set 199 srs 199 display voice sip-server register-user 200 display voice sip-server resource-statistic 201 expires 202 mode 203 number 203 probe remote-server 204 register-user 205 rule 205 service 206 server-bind ipv4 207 server enable 207 sip-server 208 trunk 209 trusted-point 209 SIP trunk configuration commands 211 address 211 address sip server-group 212 assign 212 account enable 213 bind sip-trunk account 214 codec transparent 215 description 215 display voice sip-trunk account 216 display voice server-group 217 group-name 218 hot-swap enable 219 keepalive 219 match source host-prefix 220 match destination host-prefix 221 match source address 222 proxy server-group 223 registrar server-group 223 register enable 224 redundancy mode 225 server-group 225 sip-trunk account 226 sip-trunk enable 227 user 227

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Call services configuration commands 229 backup-rule loose 229 call-forwarding no-reply enable 229 call-forwarding on-busy enable 230 call-forwarding priority 231 call-forwarding unavailable enable 231 call-forwarding unconditional enable 232 call-hold enable 233 call-hold-format 233 call-transfer enable 234 call-transfer start-delay 235 call-waiting 235 call-waiting enable 236 call-waiting priority 237 conference enable 237 dialin-restriction enable 238 dialout-restriction enable 239 display voice sip subscribe-state 239 display voice ss mwi 240 feature 242 hunt-group enable 243 hunt-group priority 243 joined-conference enable 244 mwi enable 245 mwi tone-duration 245 mwi-server 246 timer called-hookon-delay 247 Call-watch configuration commands 249 call-watch group 249 call-watch rule 250 display call-watch status 251 Fax over IP configuration commands 253 default entity fax 253 display voice fax 255 fax baudrate 258 fax cng-switch enable 259 fax ecm 259 fax level 260 fax local-train threshold 261 fax nsf-on 261 fax protocol 262 fax train-mode 263 modem compatible-param 264 modem protocol 265 reset voice fax statistics 265 IVR configuration commands 267 call-normal 267 description 268 display voice ivr call-info 268 display voice ivr media-play 269 display voice ivr media-source 270 entity ivr 271 extension 272
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input-error 273 ivr-input-error 274 ivr-root 275 ivr-system 275 ivr-timeout 276 media-file 277 media-play 277 node 278 operation 279 select-rule operation-order 280 set-media 280 timeout 281 user-input 282

VoFR configuration commands 284 address 284 call-mode 285 cid select-mode 285 display fr vofr-info 286 entity vofr 287 outband vofr 288 seq-number 288 timestamp 289 trunk-id 290 voice bandwidth 290 vofr 291 vofr frf11-timer 292 Voice RADIUS configuration commands 294 aaa-client 294 accounting 294 accounting-did 295 acct-method 296 authentication 297 authentication-did 297 authorization 298 authorization-did 299 callednumber receive-method 300 card-digit 301 cdr 301 display voice access-number 302 display voice call-history-record 305 display voice radius statistic 308 gw-access-number 310 password-digit 311 process-config 312 redialtimes 313 reset voice radius statistic 314 selectlanguage 315 timer two-stage dial-interval 316 Support and other resources 317 Contacting HP 317 Subscription service 317 Related information 317 Documents 317
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Websites 317 Conventions 318

Index 320

Voice entity configuration commands

call-history
Description
Use call-history max-count to configure the maximum number of call history records that can be stored. Use undo call-history max-count to restore the default. By default, the maximum number of call history records that can be stored is 50.

Syntax
call-history max-count number undo call-history max-count

View
Voice view

Default level
2: System level

Parameters
number: Maximum number of call history records that can be stored, in the range of 0 to 200.

Examples
# Configure the maximum number of call history records that can be stored as 100.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] call-history max-count 100

compression
Description
Use compression to specify the codecs and their priority levels for the voice entity. Use undo compression to restore the default value. By default, the codec with the first priority is g729r8, that with the second priority is g71 1alaw, that with the third priority is g71 1ulaw, and that with the fourth priority is g723r53. g71 1alaw and g71 1ulaw provide high-quality voice transmission, while requiring greater bandwidth. g723r53 and g723r63 provide silence suppression technology and comfortable 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 g729r8 and g729a 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. Table 1 describes the relationship between codec algorithms and bandwidth.
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Table 1 Relationship between algorithms and bandwidth Codec

G.711 (A-law and -law) G.726 G.729 G.723 r63 G.723 r53

Bandwidth
64 kbps (without compression) 16, 24, 32, 40 kbps 8 kbps 6.3 kbps 5.3 kbps

Voice quality
Best Good Good Fair 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 and the more bandwidth is consumed. Longer packet assembly interval results in longer fixed coding latency. The following tables show the relevant packet assembly parameters without IPHC, including packet assembly interval, bytes coded in a time unit, and network bandwidth. Thus, you can choose a suitable codec algorithm according to idle and busy status of the line and network situations more conveniently. Table 2 G.711 algorithm (A-law and -law) Packet assembly interval
10 ms 20 ms 30 ms

Bytes coded in a time unit

80 160 240

Packet length (IP) (bytes)

120 200 280

Network bandwidth (IP)

96 kbps 80 kbps 74.7 kbps

Packet length (IP+PPP) (bytes)

126 206 286

Network bandwidt h (IP+PPP)

100.8 kbps 82.4 kbps 76.3 kbps

Coding latency
10 ms 20 ms 30 ms

G.711 algorithm (A-law and -law): media stream bandwidth 64 kbps, minimum packet assembly interval 10 ms.

Table 3 G.723 r63 algorithm Packet assembly interval

30 ms 60 ms 90 ms 120 ms 150 ms 180 ms

Bytes coded in a time unit

24 48 72 96 120 144

Packet length (IP) (bytes)

64 88 112 136 160 184

Network bandwidth (IP)

16.8 kbps 11.6 kbps 9.8 kbps 9.1 kbps 8.5 kbps 8.2 kbps

Packet length (IP+PPP) (bytes)

70 94 118 142 166 190

Network bandwidth (IP+PPP)

18.4 kbps 12.3 kbps 10.3 kbps 9.5 kbps 8.9 kbps 8.4 kbps

Coding latency
30 ms 60 ms 90 ms 120 ms 150 ms 180 ms

G.723 r63 algorithm: media stream bandwidth 6.3 kbps, minimum packet assembly interval 30 ms.

Table 4 G.723 r53 algorithm Packet assembly interval

30 ms 60 ms 90 ms 120 ms 150 ms 180 ms

Bytes coded in a time unit

20 40 60 80 100 120

Packet length (IP) (bytes)

60 80 100 120 140 160

Network bandwidth (IP)

15.9 kbps 10.6 kbps 8.8 kbps 8 kbps 7.5 kbps 7.1 kbps

Packet length (IP+PPP) (bytes)

66 86 106 126 146 166

Network bandwidth (IP+PPP)

17.5 kbps 11.4 kbps 9.3 kbps 8.4 kbps 7.8 kbps 7.4 kbps

Coding latency
30 ms 60 ms 90 ms 120 ms 150 ms 180 ms

G.723 r53 algorithm: media stream bandwidth 5.3 kbps, minimum packet assembly interval 30 ms.

Table 5 G.726 r16 algorithm Packet assembly interval

10 ms 20 ms 30 ms 40 ms 50 ms 60 ms 70 ms 80 ms 90 ms 100 ms 110 ms

Bytes coded in a time unit

20 40 60 80 100 120 140 160 180 200 220

Packet length (IP) (bytes)

60 80 100 120 140 160 180 200 220 240 260

Network bandwidth (IP)

48 kbps 32 kbps 26.7 kbps 24 kbps 22.4 kbps 21.3 kbps 20.6 kbps 20 kbps 19.5 kbps 19.2 kbps 18.9 kbps

Packet length (IP+PPP) (bytes)

66 86 106 126 146 166 186 206 226 246 266

Network bandwidth (IP+PPP)

52.8 kbps 34.4 kbps 28.3 kbps 25.2 kbps 22.1 kbps 11.4 kbps 21.3 kbps 20.6 kbps 20.1 kbps 19.7 kbps 19.3 kbps

Coding latency
10 ms 20 ms 30 ms 40 ms 50 ms 60 ms 70 ms 80 ms 90 ms 100 ms 110 ms

G.726 r16 algorithm: media stream bandwidth 16 kbps, minimum packet assembly interval 10 ms.

Table 6 G.726 r24 algorithm Packet assembly interval

10 ms 20 ms 30 ms 40 ms 50 ms 60 ms

Bytes coded in a time unit

30 60 90 120 150 180

Packet length (IP) (bytes)

70 100 130 160 190 220

Network bandwidth (IP)

56 kbps 40 kbps 34.7 kbps 32 kbps 30.4 kbps 29.3 kbps

Packet length (IP+PPP) (bytes)

76 106 136 166 196 226

Network bandwidth (IP+PPP)

60.8 kbps 42.4 kbps 36.3 kbps 33.2 kbps 31.2 kbps 30.1 kbps

Coding latency
10 ms 20 ms 30 ms 40 ms 50 ms 60 ms

Packet assembly interval

70 ms

Bytes coded in a time unit

210

Packet length (IP) (bytes)

250

Network bandwidth (IP)

28.6 kbps

Packet length (IP+PPP) (bytes)

256

Network bandwidth (IP+PPP)

29.3 kbps

Coding latency
70 ms

G.726 r24 algorithm: media stream bandwidth 24 kbps, minimum packet assembly interval 10 ms.

Table 7 G.726 r32 algorithm Packet assembly interval

10 ms 20 ms 30 ms 40 ms 50 ms

Bytes coded in a time unit

40 80 120 160 200

Packet length (IP) (bytes)

80 120 160 200 240

Network bandwidth IP
64 kbps 48 kbps 42.7 kbps 40 kbps 38.4 kbps

Packet length (IP+PPP) (bytes)

86 126 166 206 246

Network bandwidth (IP+PPP)

68.8 kbps 50.4 kbps 44.3 kbps 41.2 kbps 39.4 kbps

Coding latency
10 ms 20 ms 30 ms 40 ms 50 ms

G.726 r32 algorithm: media stream bandwidth 32 kbps, minimum packet assembly interval 10 ms.

Table 8 G.726 r40 algorithm Packet assembly interval

10 ms 20 ms 30 ms 40 ms

Bytes coded in a time unit

50 100 150 200

Packet length (IP) (bytes)

90 140 190 240

Network bandwidt h (IP)

72 kbps 56 kbps 50.7 kbps 48 kbps

Packet length (IP+PPP) (bytes)

96 146 196 246

Network bandwidth (IP+PPP)

76.8 kbps 58.4 kbps 52.3 kbps 49.2 kbps

Coding latency
10 ms 20 ms 30 ms 40 ms

G.726 r40 algorithm: media stream bandwidth 40 kbps, minimum packet assembly interval 10 ms.

Table 9 G.729 algorithm Packet assembly interval

10 ms 20 ms 30 ms

Bytes coded in a time unit

10 20 30

Packet length (IP) (bytes)

50 60 70

Network bandwidth (IP)

40 kbps 24 kbps 18.7 kbps

Packet length (IP+PPP) (bytes)

56 66 76

Network bandwidth (IP+PPP)

44.8 kbps 26.4 kbps 20.3 kbps

Coding latency
10 ms 20 ms 30 ms

Packet assembly interval

40 ms 50 ms 60 ms 70 ms 80 ms 90 ms 100 ms 110 ms 120 ms 130 ms 140 ms 150 ms 160 ms 170 ms 180 ms

Bytes coded in a time unit

40 50 60 70 80 90 100 110 120 130 140 150 160 170 180

Packet length (IP) (bytes)

80 90 100 110 120 130 140 150 160 170 180 190 200 210 220

Network bandwidth (IP)

16 kbps 14.4 kbps 13.3 kbps 12.6 kbps 12 kbps 11.6 kbps 11.2 kbps 10.9 kbps 10.7 kbps 10.5 kbps 10.3 kbps 10.1 kbps 10 kbps 9.9 kbps 9.8 kbps

Packet length (IP+PPP) (bytes)

86 96 106 116 126 136 146 156 166 176 186 196 206 216 226

Network bandwidth (IP+PPP)

17.2 kbps 15.4 kbps 14.1 kbps 13.3 kbps 12.6 kbps 12.1 kbps 11.7 kbps 11.3 kbps 11.1 kbps 10.8 kbps 10.6 kbps 10.5 kbps 10.3 kbps 10.2 kbps 10 kbps

Coding latency
40 ms 50 ms 60 ms 70 ms 80 ms 90 ms 100 ms 110 ms 120 ms 130 ms 140 ms 150 ms 160 ms 170 ms 180 ms

G.729 algorithm: media stream bandwidth 8 kbps, minimum packet assembly interval 10 ms.

NOTE: Packet assembly interval is the duration to encapsulate information into a voice packet. Bytes coded in a time unit = packet assembly interval media stream bandwidth. Packet length (IP) = IP header + RTP header + UDP header + voice information length = 20+12+8+data. Packet length (IP+PPP) = PPP header + IP header + RTP header + UDP header + voice information length =
6+20+12+8+data.

Network bandwidth = Bandwidth of the media stream packet length/bytes coded in a time unit. Because IPHC compression is affected significantly by network stability, it cannot achieve high efficiency unless the line is of high quality, the network is very stable, and packet loss does not occur or seldom occurs. When the network is unstable, IPHC efficiency decreases drastically. With best IPHC performance, the IP (RTP) header can be compressed to 2 bytes. If the 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 a packet assembly interval of 30 milliseconds. Table 10 Compression efficiency of IPHC+PPP header Bytes coded in a time unit
30 24 20 60

Before compression Packet length (IP+PPP) (bytes)

76 70 66 106

After IPHC+PPP compression Packet length (IP+PPP) (bytes)

34 28 24 64

Codec

Network bandwidth (IP+PPP)

20.3 kbps 18.4 kbps 17.5 kbps 28.3 kbps 5

Network bandwidth (IP+PPP)

9.1 kbps 7.4 kbps 6.4 kbps 17.1 kbps

G.729 G.723r63 G.723r53 G.726r16

Codec

Bytes coded in a time unit

90 120 150

Before compression Packet length (IP+PPP) (bytes)

136 166 196

After IPHC+PPP compression Packet length (IP+PPP) (bytes)

94 124 154

Network bandwidth (IP+PPP)

36.3 kbps 44.3 kbps 52.3 kbps

Network bandwidth (IP+PPP)

25.1 kbps 33.1 kbps 41.1 kbps

G.726r24 G.726r32 G.726r40

Two communication parties can communicate normally only if they share some identical coding/decoding algorithms. If the codec algorithm between two connected devices is inconsistent, or the two devices do not share any common coding/decoding algorithms, the calling will fail. NOTE: For IVR voice entities, four codecs are supported: g71 1alaw, g71 1ulaw, g723r53, and g729r8. By default, the
codec with the first priority is g729r8, the codec with the second priority is g71 1alaw, the codec with the third priority is g71 1ulaw, and the codec with the fourth priority is g723r53.

The following cards support the g726 codec: the 1-port, 2-port, or 4-port FXS interface card, the 1-port, 2-port, or
4-port FXO interface card, and the 2-port or 4-port E&M interface card.

Syntax
compression { 1st-level | 2nd-level | 3rd-level | 4th-level } { g71 1alaw | g71 1ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r40 | g729a | g729br8 | g729r8 } undo compression { 1st-level | 2nd-level | 3rd-level | 4th-level }

View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view

Default level
2: System level

Parameters
1st-level: Specifies a codec with the first priority. 2nd-level: Specifies a codec with the second priority. 3rd-level: Specifies a codec with the third priority. 4th-level: Specifies a codec with the fourth priority (the lowest priority). g71 1alaw: G.71 A-law codec (defining the pulse code modulation technology), requiring a bandwidth of 1 64 kbps, usually adopted in Europe. g71 1ulaw: G.71 1-law codec, requiring a bandwidth of 64 kbps, usually adopted in North America and Japan. g723r53: G.723.1 Annex A codec, requiring a bandwidth of 5.3 kbps. g723r63: G.723.1 Annex A codec, requiring a bandwidth of 6.3 kbps. g726r16: G.726 Annex A codec. It uses the ADPCM technology, requiring a bandwidth of 16 kbps. g726r24: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 24 kbps.
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g726r32: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 32 kbps. g726r40: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 40 kbps. g729a: G.729 Annex A codec (a simplified version of G.729), requiring a bandwidth of 8 kbps. g729br8: G.729 Annex B codec. It uses CS-ACELP, requiring a bandwidth of 8 kbps. g729r8: G.729 (the voice compression technology using conjugate algebraic-code-excited linear-prediction), requiring a bandwidth of 8 kbps.

Examples
# Configure to use g723r53 coding/decoding algorithm first, then the g729r8.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] compression 1st-level g723r53 [Sysname-voice-dial-entity10] compression 2nd-level g729r8

default entity compression

Description
Use default entity compression to specify the default global codecs and their priority levels. Use undo default entity compression to restore the default. By default, the codec with the first priority is g729r8, the codec with the second priority is g71 1alaw, the codec with the third priority is g71 1ulaw, and the codec with the fourth priority is g723r53. The default entity compression command can be used to globally configure the default mode of the voice coding and decoding. After the configuration, 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 commands: compression. NOTE: The default entity compression command takes no effect on IVR voice entities.

Syntax
default entity compression { 1st-level | 2nd-level | 3rd-level | 4th-level } { g71 1alaw | g71 1ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r40 | g729a | g729br8 | g729r8 } undo default entity compression { 1st-level | 2nd-level | 3rd-level | 4th-level }

View
Voice dial program view

Default level
2: System level

Parameters
1st-level: Specifies a codec with the first priority. 2nd-level: Specifies a codec with the second priority. 3rd-level: Specifies a codec with the third priority.
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4th-level: Specifies a codec with the fourth priority (the lowest priority). g71 1alaw: G.71 A-law codec (defining the pulse code modulation technology), requiring a bandwidth of 1 64 kbps, usually adopted in Europe. g71 1ulaw: G.71 1-law codec, requiring a bandwidth of 64 kbps, usually adopted in North America and Japan. g723r53: G.723.1 Annex A codec, requiring a bandwidth of 5.3 kbps. g723r63: G.723.1 Annex A codec, requiring a bandwidth of 6.3 kbps. g726r16: G.726 Annex A codec. It uses the ADPCM technology, requiring a bandwidth of 16 kbps. g726r24: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 24 kbps. g726r32: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 32 kbps. g726r40: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 40 kbps. g729a: G.729 Annex A codec (a simplified version of G.729), requiring a bandwidth of 8 kbps. g729br8: G.729 Annex B codec. It uses CS-ACELP, requiring a bandwidth of 8 kbps. g729r8: G.729 (the voice compression technology using conjugate algebraic-code-excited linear-prediction), requiring a bandwidth of 8 kbps.

Examples
# Adopt the g723r53 coding and decoding mode as the first selection globally.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] default entity compression 1st-level g723r53

default entity payload-size

Description
Use default entity payload-size to configure the default packetization period for a codec. Use undo default entity payload-size to restore the default. Because the IVR voice entity does not support g726 codecs, the packetization periods configured for g726 codecs on an IVR voice entity take no effect. For more information about the IVR voice entity, see Voice Configuration Guide. Related commands: default entity compression, entity compression, payload-size, and set-media.

Syntax
default entity payload-size { g71 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } time-length 1 undo default entity payload-size { g71 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } 1

View
Voice dial program view

Default level
2: System level

Parameters
g71 Specifies the packetization period for g71 codec. It can be 10, 20 (the default), or 30 milliseconds. 1: 1
8

g723: Specifies the packetization period for g723 codec. It is an integral multiple of 30 in the range of 30 to 180 milliseconds. It defaults to 30 milliseconds. g726r16: Specifies the packetization period for g726r16 codec. It ranges from 10 to 1 milliseconds and 10 defaults to 30 milliseconds. g726r24: Specifies the packetization period for g726r24 codec. It ranges from 10 to 70 milliseconds and defaults to 30 milliseconds. g726r32: Specifies the packetization period for g726r32 codec. It ranges from 10 to 50 milliseconds and defaults to 30 milliseconds. g726r40: Specifies the packetization period for g726r40 codec. It ranges from 10 to 40 milliseconds and defaults to 30 milliseconds. g729: Specifies the packetization period for g729 codec. It ranges from 10 to 180 milliseconds and defaults to 30 milliseconds. time-length: Packetization period for a codec.

Examples
# Set the packetization period for G.71 codec to 30 milliseconds. 1
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] default entity payload-size g711 30

default entity vad-on

Description
Use default entity vad-on to globally configure VAD as the default value. Use undo default entity vad-on to restore the fixed value (disabling VAD) to be the default value. By default, VAD is disabled. The default entity vad-on command is used to globally enable VAD and make it as the default setting. After the configuration, all the voice entities and newly created voice entities on this router, which have not been configured with this function, will inherit this configuration (G. 71 does not support VAD). 1 Related commands: vad-on.

Syntax
default entity vad-on undo default entity vad-on

View
Voice dial program view

Default level
2: System level

Parameters
None

Examples
# Enable VAD globally.
9

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] default entity vad-on

description (voice entity view)

Description
Use description to configure a voice entity description string. Use undo description to delete the voice entity description string. By default, no description is configured for the voice entity. You can use description to add a description to a voice entity, which has no effect on the performance of the voice entity interface. You can view this description with the display command.

Syntax
description string undo description

View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view

Default level
2: System level

Parameters
string: Voice entity description string, whose length ranges from 1 to 80 characters.

Examples
# Add the description local-entity 10 to voice entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] description local-entity10

dial-trap enable
Description
Use dial-trap enable to enable the trap function for a voice entity. Use undo dial-trap enable to disable the trap function for a voice entity. By default, the trap function is disabled for a voice entity.

Syntax
dial-trap enable undo dial-trap enable

View
POTS voice entity view, VOIP voice entity view, VoFR entity view, IVR entity view
10

Default level
2: System level

Parameters
None

Examples
# Enable the trap function for VoIP voice entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] dial-trap enable

dial-program
Description
Use dial-program to enter the voice dial program view.

Syntax
dial-program

View
Voice view

Default level
2: System level

Parameters
None

Examples
# Enter the dial program view
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program

display voice call-info

Description
Use display voice call-info to display the contents in the call information table.

Syntax
display voice call-info { brief | mark tag | verbose } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

11

Parameters
brief: Displays the brief information of the call information table. mark tag: Displays the call information of the call information table by tag (in the range of 0 to 127). verbose: Displays the detailed information of the call information table. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the brief information of the call information table at a certain point of time.
<Sysname> display voice call-info brief Brief information table for current calls # **************** CALL 0 *************** ViIfIndex Module ID # End : 0x002C0060 : LGS CMC

# Display the detailed information of the call information table at a certain point of time.
<Sysname> display voice call-info verbose Detailed information table for current calls # **************** CALL 0 *************** Call direction ViIfIndex Related module ==> Module ID Module ID : LGS : CMC Reference Numbers : 1 Reference Numbers : 1 Current used voice entity : 13 Voice entities are offered : 13 # End 11 : From CS : 0x002C00F0

Table 11 Output description Field

ViIfIndex Module ID Call direction

Description
Index of the voice interface from which the call is originated ID of a voice module that the call passes through Call direction of the call

12

Field
Reference Numbers entity

Description
Number of times of referencing the call information table of a call Voice entity involved in the call.

display voice cmc

Description
Use display voice cmc to display messages which are related to the CMC module. These messages mainly contain call control block messages and statistic messages, in which statistic messages can be classified and displayed according to the type of messages and the interaction with surrounding modules.

Syntax
display voice cmc { ccb | statistic [ all | em | iva | lgs | r2 | sip | tmrout | vim ] } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
ccb: Displays the call control block of the CMC module. statistic: Displays statistics information related to the CMC module. all: Displays all statistics information related to the CMC module. em: Displays EM module information related to the CMC module. iva: Displays IVA module information related to the CMC module. lgs: Displays relevant LGS module information related to the CMC module. r2: Displays R2 module information related to the CMC module. sip: Displays SIP module information related to the CMC module. tmrout: Displays timeout information of the timer in the CMC module. vim: Displays VIM module information related to the CMC module. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the information of the call control block of the CMC module.
<Sysname> display voice cmc ccb The CMC Module Call Control Block Information! #

13

*************** CCB[1] *************** GblCallID CalledAddr CalledAddrSubst CallerAddr CallerAddrSubst Call Leg Number Active Service INCOMING LEG[0] { Spl Protocol LocalRef IfIndex IpAddress IpPort LegState ConnectState } OUTGOING CALLLEG NUMBER : 1 OUTGOING LEG[0] { Spl Protocol LocalRef IfIndex IpAddress IpPort LegState ConnectState } # End. : LGS : 0x0003 : 2884064 : 0.0.0.0 : 0 : OUT_STATE_ACTIVE : CONN_STATE_ACTIVE : LGS : 0x0002 : 2884067 : 0.0.0.0 : 0 : IN_STATE_ACTIVE : CONN_STATE_ACTIVE : 0x10000 : 2961 : 2961 : : : 2 : 0

CallInfoTabIndex : 0

INCOMING CALLLEG NUMBER : 1

# Display LGS statistics information related to the CMC module

<Sysname> display voice cmc statistic lgs ACCP Message statistics between CMC and LGS: { Send SETUP message Send SETUP_ACK message Send ALERTING message Send CONNECT message Send RELEASE message Send RELEASE_COMP message Send INFORMATION message Send SWITCH_CODEC message Send FAXVOC_SWTH message Send FAXVOC_SWTHACK message : 0 : 0 : 0 : 0 : 0 : 0 : 0 : 0 : 0 : 0

14

Receive SETUP message Receive SETUP_ACK message Receive ALERTING message Receive CONNECT message Receive RELEASE message Receive RELEASE_COMP message Receive INFORMATION message Receive SWITCH_CODEC message Receive FAXVOC_SWTH message }

: 0 : 0 : 0 : 0 : 0 : 0 : 0 : 0 : 0

Receive FAXVOC_SWTHACK message: 0

Table 12 Output description Field

GblCallID CalledAddr CalledAddrSubst CallerAddr CallerAddrSubst CallInfoTabIndex Call Leg Number Active Service Spl Protocol LocalRef IfIndex IpAddress IpPort LegState ConnectState SETUP message SETUP_ACK message ALERTING message CONNECT message RELEASE message RELEASE_COMP message INFORMATION message SWITCH_CODEC message FAXVOC_SWTH message FAXVOC_SWTHACK message

Description
Indicates the global ID of the call. Indicates the called number of the call. Indicates the called number after substitution. Indicates the caller number of the call. Indicates the caller number after substitution. Indicates the call information index of the call. Indicates the number of call legs of the call. Indicates the number of services involved in the call. Indicates the type of protocol used in the call leg. Indicates the local call identifier of the call leg. Indicates the voice interface index connected to the call leg. Indicates the IP address connected to the call leg. Indicates the port number connected to the call leg. Indicates the state of the call leg. Indicates the state of connection of the call. Statistics of SETUP messages sent to or from the LGS module Statistics of SETUP_ACK messages sent to or from the LGS module Statistics of ALERTING messages sent to or from the LGS module Statistics of CONNECT messages sent to or from the LGS module Statistics of RELEASE messages sent to or from the LGS module Statistics of RELEASE_COMP messages sent to or from the LGS module Statistics of INFORMATION messages sent to or from the LGS module Statistics of SWITCH_CODEC messages sent to or from the LGS module Statistics of FAXVOC_SWTH messages sent to or from the LGS module Statistics of FAXVOC_SWTHACK messages sent to or from the LGS module

15

display voice default all

Description
Use display voice default all to view the current default values and the system-fixed default values for voice and fax. For example, the carrier transmission energy level of GW defaults to 10 (the system-fixed default value is 15).

Syntax
display voice default all [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the current default values and the system-default values.
<Sysname> 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 -10(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) default entity fax train-mode ppp(system: ppp) default entity fax cng-switch off(system: off) 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 default entity payload-size g723 20(system: 20) 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)

16

default entity modem compatible-param 100(system: 100) default entity modem protocol pcm disable

Table 13 Output description Field

fax ecm fax protocol t38 fax redundancy t38 hb-redundancy fax redundancy t38 lb-redundancy fax level fax local-train threshold fax baudrate fax nsf-on fax train-mode fax cng-switch compression 1st-level compression 2nd-level compression 3rd-level compression 4th-level vad-on payload-size g711 payload-size g723 payload-size g726r16 payload-size g726r24 payload-size g726r32 payload-size g726r40 payload-size g729 modem compatible-param modem protocol pcm

Description
ECM mode is used for Fax. Fax protocol for intercommunication Number of high-speed redundant packets, available for standard T.38 or T.38 Number of low-speed redundant packets, available for standard T.38 or T.38 Gateway carrier transmitting energy level Fax local training threshold percentage Highest Fax rate Fax capacity negotiation mode Fax training mode CNG fax switch Voice coding mode of the first preference Voice coding mode of the second preference Voice coding mode of the third preference Voice coding mode of the fourth preference Voice entity VAD Voice entity packet assembly interval (G.711) Voice entity packet assembly interval (G.723) Voice entity packet assembly interval (G.723 r16) Voice entity packet assembly interval (G.723 r24) Voice entity packet assembly interval (G.723 r32) Voice entity packet assembly interval (G.723 r40) Voice entity packet assembly interval (G.729) Value of the payload type field for the NTE-compatible switching mode. SIP modem pass-through

display voice entity

Description
Use display voice entity to view the configuration information of voice entities. Normally speaking, you can use display current-configuration to view the information of all the active interfaces in the router as well as the global configuration information. But it will display a great deal of
17

information. So if you just want to view the configuration information of voice entities, you can use the display voice entity command.

Syntax
display voice entity { all | ivr | mark entity-tag | pots | vofr | voip } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
all: Displays all voice entities. ivr: Displays all IVR entities. mark entity-tag: Displays the voice entity specified by a tag (in the range of 1 to 2147483647). pots: Displays all POTS entities. vofr: Displays all VoFR entities. voip: Displays all VoIP entities. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the configuration information of POTS voice entities.
<Sysname> display voice entity all Current configuration of entities # entity 100 pots line 8/0 match-template 1000 # End

Table 14 Output description Field

Current configuration of entities entity 66 pots match-template line

Description
Configured voice entities POTS voice entity numbered 66 Template for number matching Voice subscriber line bound to the voice entity

18

display voice ipp statistic

Description
Use display voice ipp statistic to display statistics about the IPP module.

Syntax
display voice ipp statistic { all | cmc | h225 | h245 | ras | socket | timer } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
all: Displays all statistics about the IPP module. cmc: Displays statistics about the CMC 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. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display statistics about H.225 messages of the IPP module.
<Sysname> display voice ipp statistic h225 Statistics about H225 : { Send_Setup Send_CallProceeding Send_Alerting Send_Connect Send_ReleaseComplete Send_FacilityIndUserInput Send_FacilityTCSRequest Send_FacilityTCSAck Send_FacilityTCSReject Send_FacilityOLCRequest Send_FacilityOLCAck : : : : : : : : : : : 0 0 0 0 0 0 0 0 0 0 0

19

Send_FacilityOLCReject Send_FacilityMSDRequest Send_FacilityMSDAck Send_FacilityMSDReject Send_FacilityCLCRequest Send_FacilityCLCAck Send_FacilityStartH245 Send_Error Recv_Setup Recv_CallProceeding Recv_Alerting Recv_Connect Recv_ReleaseComplete Recv_Progress Recv_FacilityTCSRequest Recv_FacilityTCSAck Recv_FacilityTCSReject Recv_FacilityOLCRequest Recv_FacilityOLCAck Recv_FacilityOLCReject Recv_FacilityMSDRequest Recv_FacilityMSDAck Recv_FacilityMSDReject Recv_FacilityCLCRequest Recv_FacilityCLCAck Recv_Unknown }

: : : : : : : : : : : : : : : : : : : : : : : : : :

0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0

Table 15 Output description Field

Setup CallProceeding Alerting Connect ReleaseComplete FacilityIndUserInput FacilityTCSRequest FacilityTCSAck FacilityTCSReject FacilityOLCRequest FacilityOLCAck FacilityOLCReject FacilityMSDRequest

Description
Statistics of Setup messages Statistics of CallProceeding messages Statistics of Alerting messages Statistics of Connect messages Statistics of ReleaseComplete messages Statistics of UserInput messages Statistics of TCS Request messages Statistics of TCS Acknowledgement messages Statistics of TCS Reject messages Statistics of OLC Request messages Statistics of OLC Acknowledgement messages Statistics of OLC Reject messages Statistics of MSD Request messages 20

Field
FacilityMSDAck FacilityMSDReject FacilityCLCRequest FacilityCLCAck FacilityStartH245 Error Unknown

Description
Statistics of MSD Acknowledgement messages Statistics of MSD Reject messages Statistics of CLC Request messages Statistics of CLC Acknowledgement messages Statistics of H.245 Start messages Statistics of Error messages Statistics of Unknown messages

display voice iva statistic

Description
Use display voice iva statistic to view the call statistics between IVA module and other modules.

Syntax
display voice iva statistic { all | call | cmc | error | isdn | proc | timer | vim } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
all: Displays all the statistic information related to the IVA module. call: Displays the calling statistics in the IVA module. cmc: Displays all the interaction statistics between the IVA and the CMC module. error: Displays all the error statistics of the IVA module. isdn: Displays the interaction statistics between IVA module and ISDN. proc: Displays the statistic information of process call in the IVA module. timer: Displays the timers statistic information of the IVA module. vim: Displays all the interaction statistic information between IVA module and VIM. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the call statistics between IVA module and other modules.
<Sysname> display voice iva statistic call

21

Statistics about IVA calls : { IVA_ISDN_ACTIVE_CALL IVA_ISDN_ACTIVE_CALL_SUCCEEDED IVA_ISDN_ACTIVE_CALL_FAILED IVA_ISDN_PASSIVE_CALL IVA_ISDN_PASSIVE_CALL_SUCCEEDED IVA_ISDN_PASSIVE_CALL_FAILED } : : : : : : 0 0 0 0 0 0

Table 16 Output description Field

IVA_ISDN_ACTIVE_CALL IVA_ISDN_ACTIVE_CALL_SUCCEEDED IVA_ISDN_ACTIVE_CALL_FAILED IVA_ISDN_PASSIVE_CALL IVA_ISDN_PASSIVE_CALL_SUCCEEDED IVA_ISDN_PASSIVE_CALL_FAILED

Description
Statistics of calls generated when IVA serves as the caller Statistics of successful calls when IVA serves as the caller Statistics of failed calls when IVA serves as the caller Statistics of calls generated when IVA serves as the called Statistics of successful calls when IVA serves as the called Statistics of failed calls when IVA serves as the called

display voice statistics call-active

Description
Use display voice statistics call-active to view the statistics of active calls. Note the following: A call contains two directions: the incoming call and outgoing call. Therefore, two call records are generated for one call: one for the incoming call, and the other for the outgoing call. Call statistics are based on the number of call records instead of the number of calls. When multiple calls are in progress, the call records are displayed in chronological order.

Syntax
display voice statistics call-active { all | calling calling-number | called called-number } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

Parameters
all: Displays the statistics of all active calls. calling calling-number: Displays the active call statistics of the specified calling number. called called-number: Displays the active call statistics of the specified called number. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide.
22

begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the statistics of all active calls.
<Sysname> display voice statistics call-active all Current information of call active table: General Info: SetupTime:647449 ms Index:1 PhoneNumber:200 PhoneSubNumber: EntityIndex: IfIndex:0x0 ConnectTime:673269 ms CallDuration: 0 days 22h:49m:27s CallState:Active CallOrigin:Answer ChargedUnits:0 CallInfoType:speech ByteReceived:115070004 ByteTransmitted:115067526 PacketReceived:2739762 PacketTransmitted:2739703 VOIP Info: ConnectionId:0x0013 CallId:0 RemoteSignallingIPAddress:100.1.1.224 RemoteSignallingPort:5060 RemoteMediaIPAddress:100.1.1.224 RemoteMediaPort:16420 VADSwitch:0 SessionProtocol:Sipv2 CodecType:G729r8 CallingNumber:200 CalledNumber:100 SubstCallingNumber:200 SubstCalledNumber:100 General Info: SetupTime:647452 ms Index:1 PhoneNumber:100 PhoneSubNumber: EntityIndex:100

23

IfIndex:0x2c00c0 ConnectTime:673267 ms CallDuration: 0 days 22h:49m:27s CallState:Active CallOrigin:Originate ChargedUnits:0 CallInfoType:Speech ByteReceived:115068030 ByteTransmitted:115067484 PacketReceived:2739715 PacketTransmitted:2739702 PSTN Info: ConnectionId:0x0013 CallId:1 TxDuration:82191625 ms VoiceTxDuration:82191060 ms FaxTxDuration:0 ms ImgPages:0 CodecType:G729r8 CallingNumber:200 CalledNumber:100 SubstCallingNumber:200 SubstCalledNumber:100 End

Table 17 Output description Field

SetupTime Index PhoneSubNumber EntityIndex IfIndex ConnectTime

Description
The length of the time from the system starts up to the start time of the call, in milliseconds. Identification number, which defaults to 1. For the records with the same Setup Time, their index values increase by degrees. Sub-number of a phone. Not supported. Entity identification number. If the entity does not exist, the entity index is null. Index number of the interface of the voice subscriber line corresponding to the entity. Accumulated connect time to the peer since the system started up, in milliseconds. Call state:

Unknown: The call state is unknown. Connecting: A connection attempt (outgoing call) is being
CallState made.

Connected: A connection attempt (incoming call) is being

made.

Active: The call is active.

24

Field
CallOrigin ChargedUnits CallInfoType ByteReceived ByteTransmited PacketReceived PacketTransmited ConnectionId CallId RemoteSignallingIPAddress RemoteSignallingPort RemoteMediaIPAddr RemoteMediaPort SessionProtocol CallingNumber CalledNumber SubstCallingNumber SubstCalledNumber TxDuration

Description
Role in a call, originate or answer. Number of charged units for a connection; not supported. Information type for this call, Speech or Fax. Number of the received bytes. The maximum value is 4,294,967,295. Number of the transmitted bytes. The maximum value is 4,294,967,295. Number of the received packets. The maximum value is 4,294,967,295. Number of the transmitted packets. The maximum value is 4,294,967,295. Connection ID, which is used to identify a call. Identification number of the calling side. IP address of the remote signaling. Port number of the remote signaling. IP address of the remote media. Port number of the remote media. Session protocol type. Only the SIPv2 protocol is supported. Calling number before the substitution. Called number before the substitution. Substituted calling number. Substituted called number. Open duration of a call link, the open duration of the media channel, in milliseconds. Transmission duration of voice data, in milliseconds.

VoiceTxDuration

This value indicates the transmission time of data flows after the media channel is open. The general data flow, conference data flow, and fax data flow are not distinguished here. Duration of fax transmission, in milliseconds. For multiple times of fax, the values are added. Number of pages faxed. For multiple times of fax, the value is added.

FaxTxDuration

ImgPages

display voice statistics call-history

Description
Use display voice statistics call-history to view the history records of the calls that have ended. Related commands: call-history.

25

Syntax
display voice statistics call-history { all | last index } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

Parameters
all: Displays the history records of all calls that have ended. If this keyword is provided, the number of call history records can be displayed depends on the maximum number of call history records that can be stored, which is specified with the call-history command. last index: Displays the history record of the specified call that has ended. The value of the index argument ranges from 1 to 100. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the history records of all calls that have ended.
<Sysname>display voice statistics call-history all Current information of call history table: Call-History Info: Index:1 SetupTime:155451 ms PhoneNumber:7001 EntityIndex:7001 IfIndex:0x2c00f0 ConnectTime:168010 ms TerminateTime:171130 ms CallOrigin:Originate ChargedUnits:0 CallInfoType:Speech ByteReceived:18816 ByteTransmited:18816 PacketReceived:448 PacketTransmited:448 PSTN Info: ConnectionId:0x0000 CallId:1 TxDuration:65836 ms VoiceTxDuration:25280 ms FaxTxDuration:0 ms

26

ImgPages:0 CodecType:G729r8 CallingNumber:6001 CalledNumber:7001 SubstCallingNumber:6001 SubstCalledNumber:7001 Call-History Info: Index:2 SetupTime:155448 ms PhoneNumber:6001 EntityIndex:6000 IfIndex:0x0 ConnectTime:168011 ms TerminateTime:171131 ms CallOrigin:Answer ChargedUnits:0 CallInfoType:Speech ByteReceived:21798 ByteTransmited:18816 PacketReceived:519 PacketTransmited:448 VOIP Info: ConnectionId:0x0000 CallId:0 RemoteSignallingIPAddress: 100.1.1.223 RemoteSignallingPort:5060 RemoteMediaIPAddress:100.1.1.223 RemoteMediaPort:16428 VADSwitch:0 SessionProtocol:Sipv2 CodecType:G729r8 CallingNumber:6001 CalledNumber:7001 SubstCallingNumber:6001 SubstCalledNumber:7001 End

Table 18 Output description Field

SetupTime EntityIndex IfIndex

Description
Length of the time from the system starts up to the start time of the call, in milliseconds. Entity identification number. If the entity does not exist, the entity index is null. Index number of the interface of the voice subscriber line corresponding to the entity. 27

Field
ConnectTime TerminateTime CallOrigin ChargedUnits CallInfoType CallId RemoteSignallingIPAddress RemoteSignallingPort RemoteMediaIPAddr RemoteMediaPort SessionProtocol CallingNumber CalledNumber SubstCallingNumber SubstCalledNumber ConnectionId TxDuration

Description
Accumulated connect time to the peer since the system started up, in milliseconds. The length of the time from when the system starts up to when the terminate time of the call, in milliseconds. Role in a call, originate or answer. Number of charged units for a connection; not supported. Information type for this call, Speech or Fax. Identification number of the calling side. The remote signaling IP address. The remote signaling port number. The remote media IP address. The remote media port number. Session protocol type. Only SIPv2 is supported. Calling number before the substitution. Called number before the substitution. Substituted calling number. Substituted called number. Connection ID, which is used to identify a call. Open duration of a call link, the open duration of the media channel, in milliseconds. Transmission duration of voice data, in milliseconds.

VoiceTxDuration

This value indicates the transmission time of the data flow after the media channel is open. The general data flow, conference data flow, and fax data flow are not distinguished here. Duration of fax transmission, in milliseconds. For multiple times of fax, the values are added. Number of pages faxed. For multiple times of fax, the values are added.

FaxTxDuration

ImgPages

display voice statistics entity

Description
Use display voice statistics entity to view the call statistics of voice entities after the system starts up. The displayed statistics include number of successful calls, number of failed calls, number of accepted calls, number of refused calls, and the setup time of the last call. NOTE: This command does not cover IVR or VoFR voice entities.

28

Syntax
display voice statistics entity { all | mark entity-index } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

Parameters
all: Displays the call statistics of all voice entities. mark entity-index: Displays the call statistics of the specified entity. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the call statistics of all voice entities.
<Sysname> display voice statistics entity all Current statistics of all entities: Index:100 Type:pots Match-Template:100 ConnectTime:0 s SuccessfulCalls:0 FailedCalls:0 AcceptedCalls:0 RefusedCalls:0 LastSetupTime:0 ms Index:200 Type:pots Match-Template:200 ConnectTime:758 s SuccessfulCalls:0 FailedCalls:0 AcceptedCalls:1 RefusedCalls:0 LastSetupTime:6190ms End

29

Table 19 Output description Field

Index Type Match-Template ConnectTime LastSetupTime

Description
Entity index Entity type, which can be POTS, VoIP, or Other. Number template Accumulated connect time to the peer since the system started up, in milliseconds Setup time of the last call, in milliseconds

distinguish-localtalk
Description
Use distinguish-localtalk to enable the local call identification function. Use undo distinguish-localtalk to disable this function. By default, the local call identification function is disabled. NOTE: Configuring the three-party conference service in voice subscriber line view will invalidate the configuration of the distinguish-localtalk command. For more information about the three-party conference service, see Call services configuration commands.

Syntax
distinguish-localtalk undo distinguish-localtalk

View
Voice view

Default level
2: System level

Parameters
None

Examples
# Enable the local call identification function.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] distinguish-localtalk

dscp media
Description
Use dscp media to set the DSCP value in the ToS field in the IP packets that carry the RTP stream of the voice entity.
30

Use undo dscp media to restore the default DSCP. By default, the DSCP value is ef (101 10). 1

Syntax
dscp media dscp-value undo dscp media

View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view

Default level
2: System level

Parameters
dscp-value: DSCP value in the range of 0 to 63 or the keyword af1 af12, af13, af21, af22, af23, af31, af32, 1, af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, and ef.

Examples
# Set the DSCP value in the ToS field of the IP packets that carry the RTP stream of VoIP voice entity to af41.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 2 voip [Sysname-voice-dial-entity2] dscp media af41

entity
Description
Use entity to enter voice entity view, or configure a voice entity and then enter its view if the voice entity does not exist. Use undo entity to remove the existing voice entity. In a global view, use entity to enter a voice entity view, and use quit to return to the dial program view. For more information about IVR and VoFR voice entities, see Voice Configuration Guide. Related commands: line. NOTE: The entity-number assigned to a VoIP, POTS, or IVR entity must be unique among all VoIP and POTS entities. The system supports up to 1,000 voice entities.

Syntax
entity entity-number [ pots | voip ] undo entity { entity-number | all | pots | voip }

View
Voice dial program view

Default level
2: System level
31

Parameters
entity-number: Identifies a voice entity. The value ranges from 1 to 2147483647. all: All voice entities, including VoIP, POTS, VoFR, and IVR voice entities. pots: Indicates that the voice entity originates a call from the local voice subscriber line. voip: Indicates that the voice entity originates a call from the network side.

Examples
# Create and enter voice entity view to configure a POTS voice entity whose identification is 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots

line
Description
Use line to associate the voice entity with a specified voice subscriber line. Use undo line to remove this association. By default, there is no association between a voice entity and a voice subscriber line.

Syntax
line line-number undo line

View
POTS voice entity view

Default level
2: System level

Parameters
line-number: Number of a subscriber line.

Examples
# Associate voice entity 10 and voice subscriber line 1/0.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] line 1/0

match-template
Description
Use match-template to configure the number template for a voice entity. Use undo match-template to remove the configuration.

32

By default, no number template is bound to the local voice subscriber line in POTS view, no number template is configured for the terminating side when the POTS voice entity serves as a trunk, and no number template is configured for the voice entity in VoIP, VoFR, or IVR entity view. The number template defined by match-template 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 number 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. When configuring a POTS voice entity, use match-template to define the number template to be bound to the local voice entity. When configuring a VoIP or VoFR entity, use match-template to define the number template on the called side. When configuring an IVR entity, use match-template to define the IVR access number. NOTE: In E1 voice, T, #, and * are not supported at this time.

Syntax
match-template match-string undo match-template

View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view

Default level
2: System level

Parameters
match-string: Number template. Its format is [ + ] { string [ T ] [ $ ] | T }, with the maximum length of 31 characters. The characters are described in the following. +: The plus sign itself does not have special meanings. It only indicates that the following string is an effective number and the number is E.164-compliant. $: Is the last character, indicating the end of the number. That means the entire called number must match the string part before $. T: Timer. It means the system is waiting the subscriber for dialing any number till: the number length threshold is exceeded, or the subscriber inputs the terminator; or the timer expires. T is used to match a number with any digits. string: A string composed of any characters of 0123456789#*.!+%[]() -. The meanings of the characters are described in the following table:

Table 20 Meanings of the characters in string Character

0-9 # and * . !

Meaning
Numbers from 0 to 9. Each means a digit. 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. 33

Character

Meaning
The character or characters right in front of it appears once or several times. However, if a calling number starts with the plus sign, the sign itself does not have special meanings, and only indicates that the following is an effective number and the number is E.164-compliant. For example, (1) 9876(54)+ matches 987654, 98765454, 9876545454 and so on. (2) +110022 indicates +110022 is compliant with E.164. 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. For example, 9876(54)% matches 9876, 987654, 98765454, 9876545454 and so on. Select one character from the group. For example, [1-36] can match only one character among 1, 2, 3, and 6. A group of characters. For example, (123) means a string 123. It is usually used with !, %, and +. For example, 408(12)+ can match 40812 or 408121212. But it cannot match 408. That is, 12 can appear continuously and it must appear at least once.

[]

()

NOTE: 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.

If you want to use [ ] and ( ) at the same time, you must use them in the format of ( [ ] ). Other formats, such
as [ [ ] ] and [ ( ) ] are illegal.

- 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. signaling uses DTMF transmission, and since the plus sign (+) does not have a corresponding audio, the number cannot be transmitted to the called side successfully. While the DSS1 signaling uses ISDN transmission, the above problem does not exist. Therefore, you should avoid using a number that cannot be identified by the signaling itself; otherwise, the call will fail.

If a number starts with the plus sign (+), note the following when you use it on a trunk: The E&M, R2, and LGS

Examples
# Specify 5557922 as a telephone number of voice entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] match-template 5557922

# Configure a match template for VoIP voice entity 10.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 010 voip [Sysname-voice-dial-entity10] match-template 5557922

34

outband
Description
Use outband to configure out-of-band DTMF transmission. Use undo outband to restore the default. By default, the inband DTMF transmission mode is adopted. For more information about out-of-band SIP DTMF transmission mode, see Voice Configuration Guide.

Syntax
outband { nte | sip } undo outband

View
POTS/VoIP voice entity view

Default level
2: System level

Parameters
nte: Adopts DTMF named telephone event (NTE) transmission. sip: Configure the out-of-band SIP DTMF transmission mode.

Examples
# Configure the out-of-band SIP DTMF transmission for VoIP entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] address sip ip 10.1.1.2 [Sysname-voice-dial-entity10] outband sip

payload-size
Description
Use payload-size to configure the voice packetization period for different codecs. Use undo payload-size to restore the default. By default, the voice packetization period for g971 is 20 milliseconds, and that for g723, g726, and g726 is 30 milliseconds. Because the IVR voice entity does not support g726 codecs, the packetization periods configured for g726 codecs on an IVR voice entity take no effect. For more information about the IVR voice entity, see Voice Configuration Guide. Related commands: default entity compression, default entity payload-size, entity compression, and set-media.

Syntax
payload-size { g71 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } time-length 1 undo payload-size { g71 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } 1
35

View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view

Default level
2: System level

Parameters
g71 Packetization period in milliseconds for g71 1: 1alaw or g71 1ulaw codec, an integral multiple of 10 in the range of 10 to 30, with a default of 20. g723: Packetization period in milliseconds for g723r53 or g723r63 codec, an integral multiple of 30 in the range of 30 to 180, with a default of 30. g726r16: Packetization period in milliseconds for g726r16 codec, an integral multiple of 10 in the range of 10 to 1 with a default of 30. 10, g726r24: Packetization period in milliseconds for g726r24 codec, an integral multiple of 10 in the range of 10 to 70, with a default of 30. g726r32: Packetization period in milliseconds for g726r32 codec, an integral multiple of 10 in the range of 10 to 50, with a default of 30. g726r40: Packetization period in milliseconds for g726r40 codec, an integral multiple of 10 in the range of 10 to 40, with a default of 30. g729: Packetization period in milliseconds for g729r8 or g729a codec, an integral multiple of 10 in the range of 20 to 180, with a default of 30. time-length: DSP packetization period for a codec.

Examples
# Set the voice packetization period of the DSP for g71 codec to 30 milliseconds. 1
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] payload-size g711 30

register-number
Description
Use register-number to enable the VoIP gateway to register numbers of a voice entity with an SIP server. Use undo register-number to disable a gateway from registering numbers of a voice entity with an SIP server. By default, after configured with SIP-registration related parameters, a POTS voice entity initiates registration to the SIP server. In some cases, you need to configure the same POTS voice entity on multiple gateways. As a SIP server cannot have the same number, you cannot register a POTS voice entity with a SIP server at the same time. In other cases, you may need to register only some port numbers on the gateway with a SIP server to meet some special requirements. You can use undo register-number to specify the voice entity whose number does not need to be registered. Related commands: match-template.

36

Syntax
register-number undo register-number

View
POTS voice entity view, IVR entity view

Default level
2: System level

Parameters
None

Examples
# Specify the gateway not to register the numbers of POTS voice entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] undo register-number

reset voice cmc statistic

Description
Use reset voice cmc statistic to clear calling statistics on the CMC module. Related commands: display voice cmc.

Syntax
reset voice cmc statistic

View
User view

Default level
2: System level

Parameters
None

Examples
# Clear calling statistics on the CMC module.
<Sysname> reset voice cmc statistic

reset voice ipp statistic

Description
Use reset voice ipp statistic to reset IPP statistics. Related commands: display voice ipp statistic.

37

Syntax
reset voice ipp statistic

View
User view

Default level
2: System level

Parameters
None

Examples
# Clear IPP statistics.
<Sysname> reset voice ipp statistic

reset voice iva statistic

Description
Use reset voice iva statistic to clear IVA statistics. Related commands: display voice iva statistic.

Syntax
reset voice iva statistic

View
User view

Default level
2: System level

Parameters
None

Examples
# Clear IVA statistics.
<Sysname> reset voice iva statistic

rtp payload-type nte

Description
Use rtp payload-type nte to configure the payload type field in RTP packets in the case of DTMF relay using NTE. Use undo rtp payload-type nte to restore the default. By default, the payload type field in RTP packets is set to 101 in the case of DTMF relay using NTE.

38

NOTE: It is forbidden to set the NTE payload type field to 98, which has already been used to identify nonstandard T38 fax
packets.

When the device is connected with devices of other manufacturers for communication, you cannot set the payload
type field to any forbidden by these routers. Otherwise, an NTE negotiation failure may occur.

Syntax
rtp payload-type nte value undo rtp payload-type nte

View
POTS voice entity view, VoIP voice entity view

Default level
2: System level

Parameters
value: Value of the payload type field in RTP packets, in the range of 96 to 127.

Examples
# Set the NTE payload type field to 102 for VoIP voice entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] rtp payload-type nte 102

send-ring
Description
Use send-ring to enable the local end to play ringback tone. Use undo send-ring to disable the local end from playing ringback tone. By default, the local end does not play ringback tone. In VoIP view, this command is available only after the fast connection function is enabled or a SIP routing policy is configured. In POTS view, you can configure this command as long as the line line number command binds the POTS voice entity to a voice subscriber line rather than an FXS or FXO voice subscriber line.

Syntax
send-ring undo send-ring

View
POTS voice entity view, VoIP voice entity view, VoFR entity view

Default level
2: System level

39

Parameters
None

Examples
# Enable the local end to play ringback tone.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] send-ring

shutdown (voice entity view)

Description
Use shutdown to change the management status of the specified voice entity from UP to DOWN. Use undo shutdown to restore the default management status of the voice entity. By default, the voice entity management status is UP. Running shutdown will cause the voice entity unable to make calls.

Syntax
shutdown undo shutdown

View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view

Default level
2: System level

Parameters
None

Examples
# Change the management status of voice entity 4 to DOWN.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 4 pots [Sysname-voice-dial-entity4] shutdown

vad-on
Description
Use vad-on to enable VAD. Use undo vad-on to disable VAD. By default, VAD is disabled. If you execute vad-on or undo vad-on without specifying a codec, VAD for all codecs is enabled or disabled.
40

The G.71 and G.726 codecs do not support VAD. 1 The G.729br8 codec always supports VAD. The VAD discriminates between silence and speech on a voice connection according to signal energies. VAD reduces the bandwidth requirements of a voice connection by not generating traffic during periods of silence in an active voice connection. Speech signals are generated and transmitted only when an active voice segment is detected. Researches show that VAD can save the transmission bandwidth by 50%. Related commands: cng-on.

Syntax
vad-on [ g723r53 | g723r63 | g729a | g729r8 ] * undo vad-on [ g723r53 | g723r63 | g729a | g729r8 ] *

View
POTS voice entity view, VoIP voice entity view, VoFR entity view

Default level
2: System level

Parameters
g723r53: Specifies the g723r53 codec. g723r63: Specifies the g723r63 codec. g729a: Specifies the g729a codec. g729r8: Specifies the g729r8 codec.

Examples
# Enable VAD on POTS voice entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] vad-on

voice-setup
Description
Use voice-setup to enter voice view and enable voice services. Use undo voice-setup to disable voice services and quite voice view.

Syntax
voice-setup undo voice-setup

View
System view

Default level
2: System level

41

Parameters
None

Examples
# Enter voice view and enable voice services.
<Sysname> system-view [Sysname] voice-setup

voip timer
Description
Use voip timer to set the time duration for switching from the current VoIP link to another VoIP link or a PSTN link in case of a VoIP call failure. Use undo voip timer to restore the default. By default, the duration is five seconds. For more information about call backup, see Voice Configuration Guide.

Syntax
voip timer voip-to-pots time undo voip timer voip-to-pots

View
Voice view

Default level
2: System level

Parameters
voip-to-pots time: Specifies the time duration in seconds for switching from the current VoIP link to another VoIP link or a PSTN link (that is, the call backup switching time) in case of a VoIP call failure, in the range of 3 to 30.

Examples
# Set the time duration for switching from the current VoIP link to another VoIP link or a PSTN link in case of a VoIP call failure to 3 seconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] voip timer voip-to-pots 3

vqa dscp
Description
Use vqa dscp to globally set the DSCP subfield in the ToS field in IP packets that carry the RTP stream or voice signaling. Use undo vqa dscp to restore the default. By default, the DSCP subfield is set to ef, 101 10. 1

42

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.

Syntax
vqa dscp { media | signal } dscp-value undo vqa dscp { media | signal }

View
Voice view

Default level
2: System level

Parameters
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 af1 af12, af13, af21, af22, af23, af31, af32, 1, af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, or ef. Table 21 DSCP values Keyword
af11 af12 af13 af21 af22 af23 af31 af32 af33 af41 af42 af43 cs1 cs2 cs3 cs4 cs5 cs6 cs7 default

DSCP value in binary

001010 001100 001110 010010 010100 010110 011010 011100 011110 100010 100100 100110 001000 010000 011000 100000 101000 110000 111000 101110 43

DSCP value in decimal

10 12 14 18 20 22 26 28 30 34 36 38 8 16 24 32 40 48 56 46

Keyword
ef

DSCP value in binary

101110

DSCP value in decimal

46

Examples
# Set the DSCP value in the ToS field in the IP packets that carry voice signaling to af41.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] vqa dscp signal af41

vqa dsp-monitor buffer-time

Description
Use vqa dsp-monitor buffer-time to set duration of monitoring DSP buffered data. Use undo vqa dsp-monitor buffer-time to restore the default. By default, the duration of monitoring DSP buffered data is 270 milliseconds. Duration greater than 240 milliseconds is recommended because too small a duration value will result in poor voice quality in the case of severe jitter.

Syntax
vqa dsp-monitor buffer-time time undo vqa dsp-monitor buffer-time

View
Voice view

Default level
2: System level

Parameters
buffer-time time: Specifies the duration in milliseconds of monitoring DSP buffered data. The value is 0 or ranges from 180 to 480.

Examples
# Set the duration of monitoring DSP buffered data to 300 milliseconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] vqa dsp-monitor buffer-time 300

44

Voice subscriber line configuration commands

The voice subscriber line in this chapter refers to a digital or analog subscriber line, unless otherwise specified.

Analog voice subscriber line configuration commands

area
Description
Use area to configure the type of busy tone for FXO voice subscriber line. Use undo area to restore the default type. By default, the busy tone compliant with the Europe standard is used. This command applies to 2-wire loop trunk subscriber line FXO only. Once this command is configured, the configuration will be effective to all the analog FXO voice cards on the device. When an FXO interface card is connected to a common subscriber line of a program-controlled switch, if the user on the switch side hangs up first, the router can know that the user has hung up only after detecting the busy tone. This is made possible because different switches adopt different cptone schemes with varying frequency spectrum characteristics, based on which the busy tone can be identified.

Syntax
area { custom | europe | north-america } undo area

View
Voice view

Default level
2: System level

Parameters
custom: Busy tone defined by users. europe: Busy tone compliant with Europe standard. north-america: Busy tone compliant with North America standard.

Examples
# Configure the busy tone type compliant with the North America standard.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] area north-america

45

busytone-hookon timer
Description
Use busytone-hookon timer to configure the delay time before an on-hook for an FXO voice subscriber line. Use undo busytone-hookon timer to restore the default. By default, the delay time before an on-hook for an FXO voice subscriber line is 0 seconds. Usually, after the FXO interface detects a busy tone, the system automatically disconnects the call and immediately removes the connection. When an FXO subscriber line is used as the VoIP access port can cooperate with an IP phone, because the IP phone does not play any prompt tone to the IP phone user, it is easily for the IP phone user to ignore the busy tone and considers that the line failure occurs when the FXO subscriber line detects the busy tone and removes the connection quickly. With the delay time before an on-hook configured, when the FXO subscriber line detects a busy tone, it waits for a period of time, and then disconnects a call and removes the connection. In this case, the busy tone is first sent to the FXO interface and then sent to the IP phone, and the IP phone user will easily confirm the busy tone information before the connection is removed.

Syntax
busytone-hookon timer seconds undo busytone-hookon timer

View
Analog FXO voice subscriber line view

Default Level
2: System level

Parameters
seconds: Specifies delay time (in seconds) before an on-hook. The value is in the range of 0 to 30.

Examples
# Configure the delay time before an on-hook for an FXO voice subscriber line to 5 seconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] busytone-hookon timer 5

busytone-t-th
Description
Use busytone-t-th to configure the number of busy tone periods for detection. Use undo busytone-t-th to restore the default. By default, the number of busy tone periods for detection is 2. Enabling the busy tone detection is optional. Under particular situations, however, the actual busy tone data cannot exactly match the busy tone parameters configured for the system. If there is a big difference, the busy tone may not be detected correctly, resulting in on-hook failures or wrong on-hooks. By adjusting the time threshold of busy tone detection, you make the busy tone detection more precise.

46

Before you configure a threshold of busy tone detection, you must test it to make sure that on-hook operation can be done properly.

Syntax
busytone-t-th time-threshold undo busytone-t-th

View
Analog FXO voice subscriber line view

Default level
2: System level

Parameters
time-threshold: Number of busy tone periods for detection, in the range of 2 to 12. A bigger value means a longer busy tone detection time.

Examples
# Set the number of busy tone periods to 3.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] busytone-t-th 3

calling-name
Description
Use calling-name to configure the calling name. Use undo calling-name to remove the calling name. By default, no calling name is configured.

Syntax
calling-name text undo calling-name

View
Analog FXS voice subscriber line view

Default level
2: System level

Parameters
text: Name of the calling party associated with the FXS voice subscriber line, a string of 1 to 50 case-sensitive characters including numbers 0 through 9, letters A through Z or a through z, underlines (_), hyphens (-),dots (.), exclamation point (!), percent sign (%), asterisk (*), plus sign (+), grave accent (`), single quotation mark (), and tilde (~).

Examples
# Configure the calling name on the FXS voice subscriber line 1/0 as tony.
<Sysname> system-view [Sysname] subscriber-line 1/0

47

[Sysname-subscriber-line1/0] calling-name tony

cid display
Description
Use cid display to enable CID on an analog FXS voice subscriber line. The calling identity information includes the calling number and the calling name. Use undo cid display to disable CID. By default, CID is enabled on an analog FXS voice subscriber line.

Syntax
cid display undo cid display

View
Analog FXS voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable CID on voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] cid display

cid receive
Description
Use cid receive to enable CID. Use undo cid receive to disable CID. By default, CID is enabled. When CID is disabled and the calling party sends a calling number, the local FXO interface performs these actions: If a number is configured in the number 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. If wildcard dots (.) are used in the number configured in the number template for the POTS entity associated with the local FXO interface, the interface substitutes zeros for the calling numbers digits in the place of dots, for example, 1000 for 1 and then sends the substitution number to the called side.

Syntax
cid receive undo cid receive

48

View
Analog FXO voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable CID on voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] cid receive

cid ring
Description
Use cid ring to configure the time for CID check and after the CID check, the number of rings the FXO line receives before going off-hook. Use undo cid ring to restore the default. By default, CID check is performed between the first and the second rings, and the FXO line goes off-hook as soon as the check completes, that is, cid ring 1 0.

Syntax
cid ring { 0 | 1 | 2 } [ times ] undo cid ring

View
Analog FXO voice subscriber line view

Default level
2: System level

Parameters
0: CID check is performed before the phone rings. 1: CID check is performed between the first and the second rings. 2: CID check is performed between the second and the third rings. times: Ring count after the CID check before the FXO line goes off-hook. The value is in the range 0 to 5. The greater the value, the later the FXO line goes off-hook.

Examples
# Configure CID check to be performed before the phone rings on voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] cid ring 0

49

cid send
Description
Use cid send to enable the FXS or FXO voice subscriber line to send calling identity information to the remote end. Use undo cid send to disable the FXS or FXO voice subscriber line from sending calling identity information to the remote end. By default, the FXS or FXO voice subscriber line sends calling identity information to the remote end. After you configure undo cid send on the FXO voice subscriber line, the FXO voice subscriber line will not send any calling number to the called side, whether the originating side has sent it or it is configured in the number template for the voice entity associated with the FXO voice subscriber line.

Syntax
cid send undo cid send

View
Analog FXS voice subscriber line view, FXO voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Disable voice subscriber line 1/0 from sending calling identity information to the IP network.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] undo cid send

cid type
Description
Use cid type to configure the format of message (which carries the calling number information) transmitted over the FXS voice subscriber line. Use undo cid type to restore the default message format. By default, MDMF is adopted. Two formats are available: MDMF and SDMF. If the remote end supports one format only, you must use the same message format at the local end. The calling name in the calling identity information can only be transmitted in MDMF format.

Syntax
cid type { complex | simple } undo cid type

50

View
Analog FXS voice subscriber line view

Default level
2: System level

Parameters
complex: Calling identity information is transmitted in MDMF. simple: Calling identity information is transmitted in SDMF.

Examples
# Set the format of the transmitted calling identity information to SDMF on voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] cid type simple

cng-on
Description
Use cng-on to enable comfortable noise function. Use undo cng-on to disable this function. By default, the comfortable noise function is enabled. You can use this command to generate a comfortable background noise to replace the toneless intervals during a conversation. Related commands: line and vad-on.

Syntax
cng-on undo cng-on

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Disable comfortable noise function on subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] undo cng-on

51

cptone country-type
Description
CAUTION: The configuration of cptone country-type will take effect on all voice interfaces of all cards on the device. Use cptone country-type to configure the current device to play the call progress tones of a specified country or region or play the customized call progress tones. Use undo cptone country-type to restore the default. By default, China call progress tones are specified. The cptone country-type CS command enables customized call progress tones that have been set with the vi-card cptone-custom command. Related commands: vi-card cptone-custom.

Syntax
cptone country-type locale undo cptone country-type

View
Voice view

Default level
2: System level

Parameters
country-type locale: Configure the current device to play the call progress tones of a specified country or regions. 65 call progress tones are supported. Table 22 Countries or regions with supported call progress tones Code
AR AU AT BE BR BG CA CL CN CS HR CU

Country name (including customization)

Argentina Australia Austria Belgium Brazil Bulgaria Canada Chile China Customizes the call progress tones Croatia Cuba 52

Code
CY CZ DK EG FI FR DE GH GR HK HU IS IN ID IR IE IEU IL IT JP JO KE KR LB LU MO MY MX NP NL NZ NG NO PK PA

Country name (including customization)

Cyprus Czech Republic Denmark Egypt Finland France Germany Ghana Greece Hong Kong China Hungary Iceland India Indonesia Iran Ireland Ireland (UK style) Israel Italy Japan Jordan Kenya Korea Republic Lebanon Luxembourg Macau Malaysia Mexico Nepal Netherlands New Zealand Nigeria Norway Pakistan Panama

53

Code
PH PL PT RU SA SG SK SI ZA ES SE CH TH TR GB US UY ZW

Country name (including customization)

Philippines Poland Portugal Russian Federation Saudi Arabia Singapore Slovakia Slovenia South Africa Spain Sweden Switzerland Thailand Turkey United Kingdom United States Uruguay Zimbabwe

Examples
# Configure the device to play US call progress tones.
<sysname> system-view [sysname] voice-setup [sysname-voice] cptone country-type us

cptone tone-type
Description
Use cptone tone-type to configure the amplitude of the specified call progress tones. Use undo cptone tone-type to restore the default. By default, the amplitude of busy tone and congestion tone is 1000, that of dial tone and special dial tone is 400, and that of ringback tone and waiting tone is 600.

Syntax
cptone tone-type { all | busy-tone | congestion-tone | dial-tone | ringback-tone | special-dial-tone | waiting-tone } amplitude value undo cptone tone-type { all | busy-tone | congestion-tone | dial-tone | ringback-tone | special-dial-tone | waiting-tone } amplitude

54

View
Voice view

Default level
2: System level

Parameters
all: All types of call progress tones. busy-tone: Busy tone. congestion-tone: Congestion tone. dial-tone: Dial tone. ringback-tone: Ringback tone. special-dial-tone: Special dial tone. waiting-tone: Waiting tone. amplitude value: Amplitude of a progress tone, in the range of 200 to 1,500.

Examples
Set the amplitude of the busy tone to 1,200.
<sysname> system-view [sysname] voice-setup [sysname-voice] cptone tone-type busy-tone amplitude 1200

default
Description
Use default to restore the default settings for a voice subscriber line.

Syntax
default

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Restore the default settings for voice subscriber line 5/0.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] default This command will restore the default settings. Continue? [Y/N]:y

55

default subscriber-line
Description
Use default subscriber-line to configure the default receiving or transmitting gain on subscriber lines. Use undo default subscriber-line to restore the default value for all voice subscriber lines. You can use this command to increase the power of voice signal on the subscriber lines if the signal is too weak. Related commands: transmit gain and receive gain.

Syntax
default subscriber-line { receive | transmit } gain value undo default subscriber-line { receive | transmit } gain

View
Voice view

Default level
2: System level

Parameters
receive gain: Indicates the default receive gain on all subscriber lines. transmit gain: Indicates the default transmit gain on all subscriber lines. Value: Value of gain on subscriber lines, in the range of -14.0 to +13.9 dB (keeps one digit after the decimal point), and defaults to 0.

Examples
# Configure a receiving gain of 9.0 dB on all subscriber lines.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] default subscriber-line receive gain 9.0

delay hold
Description
Use delay hold to configure the delay signal duration in the delay start mode. Use undo delay hold to restore the default. By default, the delay signal duration is 400 milliseconds. Related commands: em-signal.

Syntax
delay hold milliseconds undo delay hold

View
E&M voice subscriber line view

56

Default level
2: System level

Parameters
hold milliseconds: Specifies delay signal duration (in milliseconds) in the delay start mode. The value ranges from 100 to 5,000.

Examples
# Set the delay signal duration in the delay start mode to 500 seconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] em-signal delay [Sysname-subscriber-line5/0] delay hold 500

delay rising
Description
Use delay rising to configure a delay time from when the terminating side detects a seizure signal to when it sends a delay signal in the delay start mode. Use undo delay rising to restore the default. By default, the delay time is 300 milliseconds. Related commands: em-signal.

Syntax
delay rising milliseconds undo delay rising

View
E&M voice subscriber line view

Default level
2: System level

Parameters
rising milliseconds: Specifies delay time (in milliseconds) from when the terminating side detects a seizure signal to when it sends a delay signal in the delay start mode. The value ranges from 20 to 2,000.

Examples
# Set the delay time from when the terminating side detects a seizure signal to when it sends a delay signal in the delay start mode to 700 milliseconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] em-signal delay [Sysname-subscriber-line5/0] delay rising 700

57

delay send-dtmf
Description
Use delay send-dtmf to configure a delay before the originating side sends DTMF signals in the immediate start mode. Use undo delay send-dtmf to restore the default. By default, the delay before the originating side sends DTMF signals in the immediate start mode is 300 milliseconds. Related commands: em-signal.

Syntax
delay send-dtmf milliseconds undo delay send-dtmf

View
E&M voice subscriber line view

Parameters
send-dtmf milliseconds: Specifies a delay (in milliseconds) before the originating side sends DTMF signals in the immediate start mode. The value ranges from 50 to 5,000.

Examples
# Set the delay before the originating side sends DTMF signals in the immediate start mode to 3,000 milliseconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] delay send-dtmf 3000

delay send-wink
Description
Use delay send-wink to configure an interval from when the terminating side receives a seizure signal to when it sends a wink signal in the wink start mode. Use undo delay send-wink to restore the default. By default, the interval from when the terminating side receives a seizure signal to when it sends a wink signal is 200 milliseconds in the wink start mode. Related commands: em-signal.

Syntax
delay send-wink milliseconds undo delay send-wink

View
E&M voice subscriber line view

Default level
2: System level
58

Parameters
send-wink milliseconds: Specifies an interval (in milliseconds) from when the terminating side receives a seizure signal to when it sends a wink signal in the wink start mode. The value ranges from 100 to 5,000.

Examples
# Set the interval from when the terminating side receives a seizure signal to when it sends a wink signal in the wink start mode to 700 milliseconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] em-signal wink [Sysname-subscriber-line5/0] delay send-wink 700

delay wink-hold
Description
Use delay wink-hold to configure duration the terminating side sends wink signals in the wink start mode. Use undo delay wink-hold to restore the default. By default, the duration the terminating side sends wink signals is 500 milliseconds in the wink start mode. Related commands: em-signal.

Syntax
delay wink-hold milliseconds undo delay wink-hold

View
E&M voice subscriber line view

Default level
2: System level

Parameters
wink-hold milliseconds: Specifies duration (in milliseconds) the terminating side sends wink signals in the wink start mode. The value ranges from 100 to 3,000.

Examples
# Set the duration the terminating side sends wink signals in the wink start mode to 700 milliseconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] em-signal wink [Sysname-subscriber-line5/0] delay wink-hold 700

delay wink-rising
Description
Use delay wink-rising to configure a maximum amount of time the originating side waits for a wink signal after sending a seizure signal in the wink start mode. Use undo delay wink-rising to restore the default.
59

By default, the maximum amount of time the originating side waits for a wink signal after sending a seizure signal is 3,000 milliseconds in the wink start mode. Related commands: em-signal.

Syntax
delay wink-rising milliseconds undo delay wink-rising

View
E&M voice subscriber line view

Default level
2: System level

Parameters
wink-rising milliseconds: Specifies the maximum amount of time (in milliseconds) the originating side waits for a wink signal after sending a seizure signal in the wink start mode. The value ranges from 100 to 5,000.

Examples
# Set the maximum amount of time the originating side waits for a wink signal after sending a seizure signal in the wink start mode to 2,000 milliseconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] em-signal wink [Sysname-subscriber-line5/0] delay wink-rising 2000

delay start-dial
Description
Use delay start-dial to configure the dial delay. Use undo delay start-dial to restore the default. By default, the dial delay is 1 second.

Syntax
delay start-dial seconds undo delay start-dial

View
FXS voice subscriber line view, FXO voice subscriber line view

Default level
2: System level

Parameters
seconds: Dial delay in seconds, in the range of 0 to 10.

Examples
# Set the dial delay on FXS subscriber line 1/0 to 5 seconds.
<Sysname> system-view [Sysname] subscriber-line 1/0

60

[Sysname-subscriber-line1/0] delay start-dial 5

description (voice subscriber line view)

Description
Use description to configure a subscriber line description string. Use undo description to delete the description. By default, the description for the voice subscriber line is interface-name+Interface. You can use description to add a description to a voice subscriber line, which has no effect on the performance of the voice entity. You can view this description with the display command.

Syntax
description string undo description

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice subscriber line view

Default level
2: System level

Parameters
string: Description string of voice subscriber line, whose length ranges from 1 to 80 characters.

Examples
# Mark voice subscriber line 1/0 as lab_1.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] description lab_1

disconnect lcfo
Description
Use disconnect lcfo to enable the sending of pulse signals at hangup. Use undo disconnect lcfo to disable the sending of pulse signals at hangup. By default, the sending of pulse signal at hangup is disabled, and the system plays busy tones to the other end.

Syntax
disconnect lcfo undo disconnect lcfo

View
FXS voice subscriber line view

Default Level
2: System level
61

Parameters
None

Examples
# Enable the sending of pulse signals at hangup on the FXS voice subscriber line 5/1.
<Sysname> system-view [Sysname] subscriber-line 5/1 [Sysname-subscriber-line5/1] disconnect lcfo

display voice subscriber-line

Description
Use display voice subscriber-line to view the configuration information of the subscriber line, such as the type, status, codec mode, receive and transmit gains. Related commands: subscriber-line.

Syntax
display voice subscriber-line line-number [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
line-number: Subscriber line number. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
NOTE: Actual output information may vary depending on the device model. # Display the configuration information about E&M voice subscriber line 5/0.
<Sysname> display voice subscriber-line 5/0 Current information Type Status Call Status Description Private Line Cng ----subscriber-line5/0 = Analog E&M Immediate-Start = UP = BUSYTONE = subscriber-line5/0 Interface = None = Enable

62

Echo Canceller Echo Canceller Tail-Length Nlp On Receive Gain Transmit Gain DTMF Threshold Analogue : Index Index Index Index Index Index Index Index Index Index 0 1 2 3 4 5 6 7 8 9

= Enable = 32 = Enable = 0.0 = 0.0 = 1400 = 458 = -9 = -9 = -9 = -9 = -3 = -12 = -12 = 30 = 300 = 3200 = 375 = 10 = 5 = 60 = 300 = 4-Wire = V = 0.8 db = E&M = 1 = 1 = Uninstall = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0 = 0

Index 10 Index 11 Index 12 Timer Dial-Interval Timer Wait-Digit Timer Ring-Back Delay Send-dtmf E&M Physical Wire E&M Type Slic-Gain Physical Information : Card Type Physical State Logical State Voice State ResetCount InPkts OutPkts InBytes OutBytes LastRcvPacketLen LastSndPacketLen CmdInBuff CmdInTotalBuff DataInBuff DataInTotalBuff AbortCmdCount AbortPktsCount G723R53ToR63Packet G723R63ToR53Packet ClearDspBuffCount

63

Table 23 Output description Field

Type Status Call Status Description Private-line CNG EchoCancel Nlp-on Receive gain Transmit gain DTMF Threshold Analogue Timer Dial-Interval Timer Wait-Digit Timer Ring-Back Delay Send-dtmf E&M Physical Wire E&M Type Slic-Gain Physical Information Card Type Physical State Logical State Voice State ResetCount InPkts OutPkts InBytes OutBytes LastRcvPacketLen LastSndPacketLen CmdInBuff CmdInTotalBuff AbortCmdCount AbortPktsCount

Description
Type of voice subscriber line Status of voice subscriber line Call status of voice subscriber line Description of voice subscriber line Private line dial number of voice subscriber line Comfortable noise configuration on voice subscriber line Echo duration configuration on voice subscriber line Non-linear process of echo cancel on voice subscriber line Receive gain configuration on voice subscriber line Transmit gain configuration on voice subscriber line DTMF threshold configuration of analog voice subscriber line Dial interval of voice subscriber line Period of timeout waiting for a number on voice subscriber line Period of timeout when ringing back on voice subscriber line Pre-dial delay of voice subscriber line Cable type of analog E&M voice interface Circuit type of analog E&M voice interface SLIC gain configuration of analog E&M voice interface Physical statistics information Type of the voice interface card Physical state of the voice interface Logical state of the voice interface Call state on the voice interface Indicates how many times the voice interface card is reset Number of received packets on the voice interface Number of sent packets on the voice interface Bytes of received packets on the voice interface Bytes of sent packets on the voice interface Length of the last received packet on the voice interface Length of the last sent packet on the voice interface Number of commands in the command buffer of the voice interface Total number of commands in the command buffers of the voice interface card Number of command packets discarded on the voice interface Number of packets discarded on the voice interface 64

Field
G723R53ToR63Packet G723R63ToR53Packet ClearDspBuffCount

Description
Number of G723R53 packets converted to G723R63 packets on the voice interface Number of G723R63 packets converted to G723R53 packets on the voice interface Number of DSP buffers cleared on the voice interface

dtmf amplitude
Description
Use dtmf amplitude to configure the DTMF amplitude. Once configured, the parameter applies to the whole device. Use undo dtmf amplitude to restore the default value. By default, the DTMF amplitude is 9.0 dBm. The configuration will apply to the whole device once you carry out this command.

Syntax
dtmf amplitude value undo dtmf amplitude

View
Voice view

Default level
2: System level

Parameters
value: DTMF amplitude in 0.1 dBm increments, in the range of 9.0 to 7.0.

Examples
# Configure the DTMF amplitude to 8.0 dBm.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dtmf amplitude -8.0

dtmf sensitivity-level
Description
Use dtmf sensitivity-level to set the DTMF detection sensitivity level and the absolute frequency deviation when the DTMF detection sensitivity level is set to medium. Use undo dtmf sensitivity-level to restore the default detection sensitivity level. By default, the DTMF detection sensitivity level is low. The following table shows the command and router compatibility:

65

Command
dtmf sensitivity-level

A-MSR900

A-MSR20-1X

A-MSR20

A-MSR30

A-MSR50

The following voice modules support the medium keyword:

SIC-2FXS1FXO MIM-16FXS

Syntax
dtmf sensitivity-level { high | low | medium [ frequency-tolerance value ] } undo dtmf sensitivity-level

View
Analog FXS voice subscriber line view, analog FXO voice subscriber line view

Default level
2: System level

Parameters
high: Sets the DTMF detection sensitivity level to high. In this mode, the reliability is low and detection errors may occur. low: Sets the DTMF detection sensitivity level to low. In this mode, the reliability is high, but DTMF tones may fail to be detected. medium: Sets the DTMF detection sensitivity level to medium. Support for this keyword varies with installed cards. frequency-tolerance value: Absolute frequency deviation (in percentage) when the DTMF detection sensitivity level is set to medium. The value is in the range 1.0 to 5.0 and defaults to 2.0. The greater the value, the higher the probability of false detection.

Examples
# Set the DTMF detection sensitivity level of voice subscriber line 1/0 to high.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] dtmf sensitivity-level high

dtmf time
Description
Use dtmf time to configure the related time parameters of DTMF. Use undo dtmf time to restore the default. By default, the persisting time of sending DTMF and the interval for sending DTMF are both 120 milliseconds. The configuration will apply to the whole interface once you carry out the command.

Syntax
dtmf time { interval | persist } milliseconds undo dtmf time { interval | persist }

View
Voice view
66

Default level
2: System level

Parameters
persist: Specifies the persisting time of sending DTMF. Interval: Specifies the interval for sending DTMF. milliseconds: Time in milliseconds, in the range of 50 to 500.

Examples
# Set the persisting time of sending DTMF digits to 200 milliseconds, and the interval to 300 milliseconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dtmf time persist 200 [Sysname-voice] dtmf time interval 300

dtmf threshold
Description
Use dtmf threshold to configure the sensitivity of DTMF digit detection. Use undo dtmf threshold 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 1,400, 458, -9, -9, -9, -9, -3, -12, -12, 30, 300, 3,200, 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 adopted.

Syntax
dtmf threshold analog index value undo dtmf threshold analog index

View
Analog FXS voice subscriber line view, analog FXO voice subscriber line view, analog E&M voice subscriber line view

Default level
2: System level

Parameters
analog: Analog voice subscriber line. index: Index number corresponding to a threshold, an integer 0 through 12. value: Threshold corresponding to the specified index. The value range varies with indexes. For details, see Table 24. According to the energy level of the row and column frequencies as well as the energy level of their double frequencies, the system determines whether the input DTMF digit is valid. The maximum energy of the input signal in the row frequency group is ROWMAX and the corresponding double frequency energy is ROW2nd. The maximum energy in the column frequency group is COLMAX and the corresponding double frequency energy is COL2nd.
67

Table 24 Meaning of the index numbers Index

0

Meaning
Lower limit of (ROWMAX + COLMAX). The input signal which is otherwise regarded too weak is recognized as a DTMF digit when ROWMAX + COLMAX) > 0. Upper limit of the maximum value of ROWMAX or COLMAX, whichever is larger. This limit is used for detecting the inter-digit delay. A detected digit is regarded ended only when max (ROWMAX, COLMAX) < 1. Lower limit of COLMAX/ROWMAX, where ROWMAX < COLMAX. An input signal is recognized as a DTMF digit only when 10 x (COLMAX/ROWMAX) > 2. Lower limit of ROWMAX/COLMAX when COLMAX ROWMAX. The function is similar to that of index 2. An input signal is recognized as a DTMF digit only when 10 x (ROWMAX/COLMAX) > 2. Upper limit of the ratio of the second largest energy level from the row frequency group to ROWMAX. The ratio must be lower than this limit for the input signal to be recognized as a DTMF digit. Upper limit of the ratio of the second largest energy level from the column frequency group to COLMAX. The ratio must be lower than this limit for the input signal to be recognized as a DTMF digit. Upper limit of ROW2nd/ROWMAX. An input signal is recognized as a DTMF digit only when ROW2nd/ROWMAX < 6. Upper limit of COL2nd/COLMAX. The ratio must be lower than this limit for the input signal to be recognized as a DTMF digit. Upper limit of the ratio of the maximum energy level of two extra specified frequency points to max (ROWMAX, COLMAX). The ratio must be greater than this upper limit for the input signal to be recognized as a DTMF digit. Lower limit 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.

Value range
1 to 4,999, with a default of 1,400

Remarks
The larger the value is, the higher the detection reliability is. However, the sensitivity decreases. The smaller the value is, the higher the detection reliability is. However, the sensitivity decreases. The larger the value is, the higher the detection reliability is. However, the sensitivity decreases. The smaller the value is, the higher the detection reliability is. However, the sensitivity decreases. The smaller the value is, the higher the detection reliability is. However, the sensitivity decreases. The smaller the value is, the higher the detection reliability is. However, the sensitivity decreases. The smaller the value is, the higher the detection reliability is. However, the sensitivity decreases. The smaller the value is, the higher the detection reliability is. However, the sensitivity decreases. The smaller the value is, the higher the detection reliability is. However, the sensitivity decreases. The larger the value is, the higher the detection reliability is. However, the sensitivity decreases.

1 to 4,999, with a default of 458

18 to 3 dB, with a default of 9 dB

18 to 3 dB, with a default of 9 dB

18 to 3 dB, with a default of 9 dB

18 to 3 dB, with a default of 9 dB

18 to 3 dB, with a default of 3 dB

18 to 3 dB, with a default of 12 dB

18 to 3 dB, with a default of 12 dB

30 to 150 milliseconds, with a default of 30 milliseconds

68

Index

Meaning
Frequency of the first extra frequency point specified for detection.

Value range
300 to 3,400 Hz, with a default of 300 Hz

Remarks

10

In addition, it must be a frequency 100 Hz greater than or less than the row and column frequency groups. Frequency of the second extra frequency point specified for detection.

11

In addition, it must be a frequency 100 Hz greater than or less than the row and column frequency groups. Lower limit 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.

300 to 3,400 Hz, with a default of 3,200 Hz

12

0 to 700, with a default of 375

The larger the value is, the higher the detection reliability is. However, the sensitivity decreases.

Examples
# Set the DTMF threshold 9 for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] dtmf threshold analog 9 40

echo-canceller
Description
Use echo-canceller to enable echo cancellation and set the echo duration. Use undo echo-canceller to disable the EC function. By default, the EC function is enabled. Related commands: subscriber-line and echo-canceller parameter. NOTE: The echo-canceller tail-length command is applicable only after the echo-canceller enable command is executed.

Syntax
echo-canceller { enable | tail-length milliseconds } undo echo-canceller { enable | tail-length }

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice subscriber line view

Default level
2: System level

Parameters
enable: Enables the echo cancellation (EC) function.
69

tail-length milliseconds: Echo duration in milliseconds, that is, the time that elapses from when a subscriber speaks to when the subscriber hears the echo. It ranges from 0 to 64, with a default of 0.

Examples
Configure the echo duration on voice subscriber line 1/0 to 24 milliseconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] echo-canceller enable [Sysname-subscriber-line1/0] echo-canceller tail-length 24

echo-canceller parameter
Description
Use echo-canceller parameter to configure echo cancellation parameters. Use undo echo-canceller parameter to restore the default. By default, the convergence rate of comfort noise amplitude is 0, the maximum amplitude of comfort noise is 256, the comfort noise mixture proportion control factor is 100, and the threshold of two-way talk is 1. Related commands: echo-canceller.

Syntax
echo-canceller parameter { convergence-rate value | max-amplitude value | mix-proportion-ratio value | talk-threshold value } undo echo-canceller parameter { convergence-rate | max-amplitude | mix-proportion-ratio | talk-threshold }

View
Voice view

Parameters
convergence-rate value: Sets the convergence rate of comfort noise amplitude. It ranges from 0 to 51 The 1. greater the value, the quicker the convergence. max-amplitude value: Sets the maximum amplitude of comfort noise. It ranges from 0 to 2,048. The higher the value, the greater the maximum noise amplitude. The value 0 indicates that the system performs only nonlinear processing and does not add comfort noise. mix-proportion-ratio value: Sets the comfort noise mixture proportion control factor. It ranges from 0 to 3,000 and defaults to 100. The greater the value, the higher the proportion of noise in the hybrid of noise and voice. talk-threshold value: Sets the threshold of two-way talk. It ranges from 0 to 2.

Examples
# Set the convergence rate of comfort noise amplitude to 50.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] echo-canceller parameter convergence-rate 50

70

em-phy-parm
Description
Use em-phy-parm to configure a wire scheme for the analog E&M subscriber line. Use undo em-phy-parm to restore the default. By default, the 4-wire analog E&M cable is selected. This command is only applicable only to the analog E&M subscriber line. The configuration will apply to all E&M interfaces of the card after you configure this command.

Syntax
em-phy-parm { 2-wire | 4-wire } undo em-phy-parm

View
Analog E&M voice subscriber line view

Default level
2: System level

Parameters
2-wire: Chooses the 2-wire analog E&M cable. 4-wire: Chooses the 4-wire analog E&M cable.

Examples
# Choose the 2-wire scheme for analog E&M subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] em-phy-parm 2-wire

em-signal
Description
Use em-signal to configure a start mode for an analog E&M voice subscriber line. Use undo em-signal to restore the default start mode. By default, the immediate start mode is selected for the analog E&M subscriber line.

Syntax
em-signal { delay | immediate | wink } undo em-signal

View
Analog E&M voice subscriber line view

Default level
2: System level

71

Parameters
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. 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. 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.

Examples
# Configure delay mode for E&M voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] em-phy-parm 4-wire

em-passthrough
Description
Use em-passthrough to enable E&M analog control signals pass-through. Use undo em-passthrough to disable E&M analog control signals pass-through. By default, E&M analog control signals pass-through is disabled.

Syntax
em-passthrough undo em-passthrough

View
Analog E&M voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable E&M analog control signals pass-through for E&M voice subscriber line 6/0.
<Sysname> system-view [Sysname] subscriber-line 6/0 [Sysname-subscriber-line6/0] em-passthrough

hookoff-mode
Description
Use hookoff-mode to configure the off-hook mode for the FXO voice subscriber line. Use undo hookoff-mode to restore the default. By default, the FXO voice subscriber line operates in the immediate off-hook mode.
72

Syntax
hookoff-mode { delay | immediate } undo hookoff-mode

View
Analog FXO voice subscriber line view

Default level
2: System level

Parameters
delay: Specifies the FXO voice subscriber line to operate in the delay off-hook mode. immediate: Specifies the FXO voice subscriber line to operate in the immediate off-hook mode.

Examples
# Specify an FXO voice subscriber line to operate in the delay off-hook mode.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line 1/0] hookoff-mode delay

hookoff-mode delay bind

Description
Use hookoff-mode delay bind to bind an FXS voice subscriber line to the FXO voice subscriber line. Use undo hookoff-mode delay bind to remove the binding. By default, no FXS voice subscriber line is bound to the FXO voice subscriber line. After an FXS voice subscriber line is bound to the FXO voice subscriber line, the off-hook/on-hook state of these two lines will be consistent. NOTE: To keep the consistent off-hook/on-hook state between the bound FXS and FXO voice subscriber lines, you must
consider the configurations of the private-line and caller-permit commands when executing the hookoff-mode delay bind fxs_subscriber_line command. The FXS voice subscriber line specified by fxs_subscriber_line must be the one to which the dedicated line number points. In addition, only the bound FXS voice subscriber line is allowed to originate calls to the FXO voice subscriber line by restricting incoming calls. For more information about the private-line and caller-permit command, see the chapter Dial plan configuration commands.

The bound FXS and FXO voice subscriber lines must come from the same device. Use ring-immediately keyword to quicken ringing synchronization between the FXO voice subscriber line and its
bound FXS voice subscriber line. However, for the telephone supporting calling identification display, the calling number will be displayed after the second ringing tone.

Syntax
hookoff-mode delay bind fxs_subscriber_line [ ring-immediately ] undo hookoff-mode delay bind

View
Analog FXO voice subscriber line view
73

Default level
2: System level

Parameters
fxs_subscriber_line: FXS voice subscriber line bound to the FXO voice subscriber line. ring-immediately: Specifies the immediate ringing mode.

Examples
# Specify the delay off-hook mode for the FXO voice subscriber line and bind FXS voice subscriber line 3/0 to the FXO voice subscriber line.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] hookoff-mode delay bind 3/0

hookoff-time
Description
Use hookoff-time to configure the on-hook timer length. Use undo hookoff-time to restore the default on-hook timer length. By default, no on-hook timer length is set.

Syntax
hookoff-time time undo hookoff-time

View
Analog FXO voice subscriber line view

Default level
2: System level

Parameters
time: Length of the on-hook timer in seconds, in the range of 60 to 36,000.

Examples
# Set the on-hook timer length to 500 seconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] hookoff-time 500

impedance
Description
Use impedance to set the current electrical impedance on an FXO or FXS voice subscriber line. Use undo impedance to restore the default. By default, the electrical impedance on the FXO or FXS voice subscriber line is the impedance value corresponding to China.
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Each country corresponds to an impedance value. Thus, you can specify an impedance value by specifying a country. You may just input the leading letters that uniquely identify a country without inputting a complete country name, however.

Syntax
impedance { country-name | R550 | R600 | R650 | R700 | R750 | R800 | R850 | R900 | R950 } undo impedance

View
Analog FXO voice subscriber line view, analog FXS voice subscriber line view

Default level
2: System level

Parameters
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. 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.

Examples
# Set the current electric impedance to r600 on voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] impedance r600

nlp-on
Description
Use nlp-on to enable the EC nonlinear processing function on a voice interface. Use undo nlp-on to disable the function. By default, the EC nonlinear processing function is enabled. The following table shows the command and router compatibility:

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Command

A-MSR900

A-MSR20-1X

A-MSR20

A-MSR30

A-MSR50

The following voice modules support this command: SIC-2FXS1FXO MIM-16FXS FIC-24FXS SIC-2BSV MIM-4BSV SIC-1VE1 SIC-1VT1 MIM-1VE1 MIM-1VT1 MIM-2VE1 MIM-2VT1 FIC-1VE1 FIC-1VT1 FIC-2VE1 FIC-2VT1

nlp-on

NOTE: This command takes effect only after the echo-canceller enable command is configured.

Syntax
nlp-on undo nlp-on

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Disable the EC nonlinear processing function on voice interface.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line 1/0] undo nlp-on

open-trunk
Description
Use open-trunk to enable E&M non-signaling mode. Use undo open-trunk to disable the E&M non-signaling mode. By default, the E&M non-signaling mode is disabled.
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Syntax
open-trunk { caller monitor interval | called } undo open-trunk

View
E&M voice subscriber line view

Default level
2: System level

Parameters
caller monitor interval: Enables the local voice gateway to use E&M non-signaling mode when serving as the calling side and specifies the monitoring interval in the range 60 to 600 seconds. called: Enables the local voice gateway to use E&M non-signaling mode when serving as the called side.

Examples
# Configure the PLAR function on voice subscriber line 5/0 so that 100 is automatically dialed out when the subscriber picks up the phone. Enable E&M non-signaling mode on the calling voice gateway and specify the monitoring interval as 120 seconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] private-line 100 [Sysname-subscriber-line5/0] open-trunk caller monitor 120

plc-mode
Description
Use plc-mode to configure a packet loss compensation mode for the analog FXS/FXO voice subscriber line. Use undo plc-mode to restore the default. By default, the gateway-specific algorithm is used for packet loss compensation.

Syntax
plc-mode { general | specific } undo plc-mode

View
Analog FXS voice subscriber line view, analog FXO voice subscriber line view

Default level
2: System level

Parameters
general: Uses the universal frame erasure algorithm. specific: Uses the specific algorithm provided by the voice gateway.

Examples
# Configure the voice gateway to use the universal packet loss compensation algorithm.
<Sysname> system-view [Sysname] subscriber-line 1/0

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[Sysname-subscriber-line1/0] plc-mode general

receive gain
Description
CAUTION: Gain adjustment may lead to call failures. You are not recommended to adjust the gain. If necessary, do it under the guidance of technical personnel. Use receive gain to set the gain value at the voice subscriber line input end. Use undo receive gain to restore the default. By default, the input gain on the voice interface is 0 dB. This command is applicable to FXO, FXS, analog E&M, BSV and E1/T1 voice subscriber lines. When the voice signals on the line attenuate to a relatively great extent, this command can be used to appropriately increase the voice input gain. Related commands: transmit gain and subscriber-line.

Syntax
receive gain value undo receive gain

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice subscriber line view

Default level
2: System level

Parameters
value: Voice input gain in dB, in the range of -14.0 to +13.9 with one digit after the decimal point.

Examples
# Set the voice input gain to 3.5 dB on subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] receive gain 3.5

reset voice cmc statistic

Description
Use reset voice cmc statistic to clear calling statistics on the CMC module. Related commands: display voice cmc.

Syntax
reset voice cmc statistic

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View
User view

Default level
2: System level

Parameters
None

Examples
# Clear calling statistics on the CMC module.
<Sysname> reset voice cmc statistic

reset voice ipp statistic

Description
Use reset voice ipp statistic to reset IPP statistics. Related commands: display voice ipp statistic.

Syntax
reset voice ipp statistic

View
User view

Default level
2: System level

Parameters
None

Examples
# Clear IPP statistics.
<Sysname> reset voice ipp statistic

reset voice iva statistic

Description
Use reset voice iva statistic to clear IVA statistics. Related commands: display voice iva statistic.

Syntax
reset voice iva statistic

View
User view

Default level
2: System level

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Parameters
None

Examples
# Clear IVA statistics.
<Sysname> reset voice iva statistic

ring-detect debounce
Description
Use ring-detect debounce to configure the debounce time of ring detection on a FXO subscriber line. By setting different debounce times, you can detect ring signals of different frequencies and waveforms. Use undo ring-detect debounce to restore the default. By default, the debounce time is 10 milliseconds. The following table shows the command and router compatibility: Command A-MSR900
SIC-2FXO SIC-1FXO MIM-4FXO MIM-2FXO FIC-4FXO

A-MSR20-1X

A-MSR20

A-MSR30

A-MSR50

The following voice modules support this command:

ring-detect debounce

NOTE: Do not set the debounce time during a conversation. You are recommended not to set a very short debounce time, because when there is line interference, short
debounce time may cause misdetection.

If you configure this command on a FXO voice subscriber line of a board, the configuration is effective for all FXO
subscriber lines of this board.

Syntax
ring-detect debounce value undo ring-detect debounce

View
Analog FXO voice subscriber line view

Default level
2: System level

Parameters
value: Debounce time of ring detection, in milliseconds, in the range of 4 to 15.

Examples
# Configure the debounce time of ring detection on FXO voice subscriber line 1/0 to 15 milliseconds.
<sysname> system-view [sysname] subscriber-line 1/0

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[sysname-subscriber-line1/0] ring-detect debounce 15

ring-detect frequency
Description
Use ring-detect frequency to set the frequency value in ring detection. Use undo ring-detect frequency to restore the default. By default, the frequency in the ring detection is 40 Hz. The following table shows the command and router compatibility: Command
ring-detect frequency

A-MSR900

A-MSR20-1X

A-MSR20

A-MSR30

A-MSR50

The following voice modules support this command:

SIC-2FXS1FXO

Syntax
ring-detect frequency value undo ring-detect frequency

View
Analog FXO voice subscriber line view

Default level
2: System level

Parameters
value: Frequency value in the ring detection, in Hz. The value is in the range 30 to 100 with the step of 10.

Examples
# Set the frequency value in ring detection on FXO voice subscriber line 1/0 to 100 Hz.
<sysname> system-view [sysname] subscriber-line 1/0 [sysname-subscriber-line1/0] ring-detect frequency 100

send-busytone
Description
Use send-busytone to enable busy tone sending on the FXO interface. Use undo send-busytone to disable busy tone sending on the FXO interface. By default, busy tone sending is disabled.

Syntax
send-busytone { enable | time seconds } undo send-busytone { enable | time }

View
Analog FXO voice subscriber line view

81

Default level
2: System level

Parameters
enable: Enables busy-tone sending on the FXO interface. time seconds: Duration of busy tone in seconds, in the range of 2 to 15. It defaults to 3 seconds. This parameter is not available without using send-busytone enable to enable busy-tone sending function.

Examples
# Enable FXO interface 1/0 to send busy tone that lasts 5 seconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] send-busytone enable [Sysname-subscriber-line1/0] send-busytone time 5

shutdown (voice subscriber line view)

Description
Use shutdown to set the voice subscriber line DOWN. Use undo shutdown to restore the default status of the voice subscriber line. By default, the voice subscriber line is UP. The POTS interface on the voice interface card will be DOWN and there will be no sound on the connected telephone after shutdown is executed, and whereas the specified voice subscriber line will be UP after undo shutdown is executed.

Syntax
shutdown undo shutdown

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Shut down voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] shutdown

82

silence-th-span
Description
Use silence-th-span to set the silence duration for automatic on-hook. Use undo silence-th-span to restore the default. By default, the silence threshold is 20 and the silence duration for automatic on-hook is 7,200 seconds (2 hours).

Syntax
silence-th-span threshold time-length undo silence-th-span

View
Analog FXO subscriber line view

Default level
2: System level

Parameters
threshold: Silence threshold. If the amplitude of voice signals from the switch is smaller than this value, the system regards the voice signals as silence. This threshold ranges from 0 to 200. Normally, the signal amplitude on the links without traffic is in the range of 2 to 5. time-length: Silence duration for automatic on-hook. Upon expiration of this duration, the system performs on-hook automatically. It ranges from 2 to 7,200 seconds.

Examples
# Set the silence threshold to 20 and the silence duration to 10 seconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] silence-th-span 20 10

slic-gain
Description
Use slic-gain to configure the output gain of the SLIC chip. The bottom layer tunes the signal gain through the SLIC chip. Use undo slic-gain to restore the default output gain. By default, the output gain of the SLIC chip is 0 dB.

Syntax
slic-gain { 0 | 1 } undo slic-gain

View
Analog E&M voice subscriber line view

Default level
2: System level
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Parameters
0: Sets the output gain of the SLIC chip to 0.8 dB. 1: Sets the output gain of the SLIC chip to 2.1 dB.

Examples
# Set SLIC-gain to 1 in analog E&M voice subscriber line view.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] slic-gain 1

subscriber-line
Description
Use subscriber-line to enter the specified voice subscriber line view. Use subscriber-line line-number to enter the voice subscriber line view. For example, if line-number is an FXS voice subscriber line, the system will enter the FXS voice subscriber line view; if line-number is an analog E&M voice subscriber line, the system will enter analog E&M voice subscriber line view.

Syntax
subscriber-line line-number

View
System view

Default level
2: System level

Parameters
line-number: Voice subscriber line number.

Examples
# Enter the view of the voice subscriber line 1/0 in system view.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0]

timer dial-interval
Description
Use timer dial-interval to configure the maximum interval for dialing the next digit. Use undo timer dial-interval to restore the default setting. By default, the maximum interval for dialing the next digit is 10 seconds. This timer will restart each time the subscriber dials a digit and will work in this way until all the digits of the number are dialed. If the timer expires before the dialing is completed, the subscriber will be prompted to hook up and the call is terminated.

Syntax
timer dial-interval seconds
84

undo timer dial-interval

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view

Default level
2: System level

Parameters
seconds: Maximum interval in seconds for dialing the next digit, in the range of 1 to 300.

Examples
# Set the maximum duration waiting for the next digit on voice line 1/0 to 5 seconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer dial-interval 5

timer disconnect-pulse
Description
Use timer disconnect-pulse to configure the time duration for the sending of the pulse signals at hangup. Use undo timer disconnect-pulse to restore the default setting.

Syntax
timer disconnect-pulse milliseconds undo timer disconnect-pulse

View
FXS voice subscriber line view

Default Level
2: System level

Parameters
milliseconds: Time duration (in milliseconds) for the sending of the pulse signals at hangup. The value is in the range 0 to 1500, and defaults to 750.

Examples
# Configure the time duration for the sending of the pulse signals at hangup on the FXS voice subscriber line 5/1 as 1 second.
<Sysname> system-view [Sysname] subscriber-line 5/1 [Sysname-subscriber-line5/1] timer disconnect-pulse 1000

timer first-dial
Description
Use timer first-dial to configure the maximum interval between off-hook and dialing the first digit. Use undo timer first-dial to restore the default setting.
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By default, the maximum interval between off-hook and dialing the first digit is 15 seconds. Upon the expiration of the timer, the subscriber will be prompted to hook up and the call is terminated.

Syntax
timer first-dial seconds undo timer first-dial

View
FXS voice subscriber line view, FXO voice subscriber line view

Default level
2: System level

Parameters
seconds: Maximum interval in seconds between off-hook and dialing the first digit, in the range of 1 to 300.

Examples
# Set the maximum interval between off-hook and dialing the first digit to 10 seconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer first-dial 15

timer hookflash-detect
Description
Use timer hookflash-detect to configure the time range for the duration of an on-hook condition that will be detected as a hookflash. Use undo timer hookflash-detect to restore the default. By default, the time range is 50 to 180 milliseconds, that is, if an on-hook condition that lasts for a period that falls within the hookflash duration range is considered a hookflash.

Syntax
timer hookflash-detect hookflash-range undo timer hookflash-detect

View
Analog FXS subscriber line view

Default level
2: System level

Parameters
hookflash-range: Hookflash duration range, in milliseconds, in the range of 50 to 1,200.

Examples
# Set the hookflash duration range for voice subscriber line 1/0 to 100 to 200 milliseconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer hookflash-detect 100-200

86

timer hookoff-interval
Description
Use timer hookoff-interval to configure the interval between on-hook and off-hook. Use undo timer hookoff-interval to restore the default. By default, the interval between on-hook and off-hook is 500 milliseconds. In the delay off-hook mode, the on-hook/off-hook state of FXS and FXO voice subscriber lines is consistent. When an FXS voice subscriber line goes off-hook, the FXO voice subscriber line to which the FXS voice subscriber line is bound goes off-hook, too. When the FXS voice subscriber line in the off-hook state needs to connect the FXO voice subscriber line to originate a call over PSTN, the FXO voice subscriber line must first perform an on-hook operation, and then perform an off-hook operation to send the called number. Related commands: hookoff-mode.

Syntax
timer hookoff-interval milliseconds undo timer hookoff-interval

View
Analog FXO voice subscriber line view

Default level
2: System level

Parameters
milliseconds: Interval between on-hook and off-hook in milliseconds, in the range of 500 to 4,000.

Examples
# Set the interval from on-hook to off-hook for FXO voice subscriber line 1/0 to 600 milliseconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer hookoff-interval 600

timer ring-back
Description
Use timer ring-back to configure the maximum duration of playing the ringback tone. Use undo timer ring-back to restore the default. By default, the maximum duration of playing the ringback tone is 60 seconds.

Syntax
timer ring-back seconds undo timer ring-back

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view

87

Default level
2: System level

Parameters
seconds: Maximum duration in seconds of playing ringback tone, in the range of 5 to 120.

Examples
# Set the maximum time duration of playing ringback tones to eight seconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer ring-back 8

timer wait-digit
Description
Use timer wait-digit to configure the maximum time duration the system waits for a digit. Use undo timer wait-digit to restore the default time settings. By default, the maximum time duration the system waits for a digit is 5 seconds.

Syntax
timer wait-digit { seconds | infinity } undo timer wait-digit

View
E&M voice subscriber line view

Default level
2: System level

Parameters
seconds: Maximum duration in seconds the system waits for a digit, in the range of 3 to 600. infinity: Infinite time.

Examples
# Set the maximum duration waiting for the first dial on voice line 5/0 to 5 seconds.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] timer wait-digit 5

transmit gain
Description
CAUTION: Gain adjustment may lead to call failures. You are not recommended to adjust the gain. If necessary, do it under the guidance of technical personnel. Use transmit gain to set the voice subscriber line output end gain value.
88

Use undo transmit gain to restore the default value. By default, the output gain on the voice interface is 0 dB. This command is applicable to FXO, FXS, E&M, BSV and E1/T1 voice subscriber lines. When a relatively small voice signal power is needed on the output line, this command can be used to properly increase the voice output attenuation value. Related commands: receive gain and subscriber-line.

Syntax
transmit gain value undo transmit gain

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice subscriber line view

Default level
2: System level

Parameters
value: Voice output gain in dB, in the range of -14.0 to 13.9 with one digit after the decimal point.

Examples
# Set the voice output gain value to 6.7dB on subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] transmit gain -6.7

type
Description
Use type to configure the analog E&M subscriber line signal type. Use undo type to cancel the existing settings. By default, the analog E&M subscriber line signal type is type 5. This command is only applicable to an E&M subscriber line, and once configured, is effective on all analog E&M lines in the corresponding slot.

Syntax
type { 1 | 2 | 3 | 5 } undo type

View
Analog E&M voice subscriber line view

Default level
2: System level

Parameters
1, 2, 3 and 5: Correspond respectively to the four signal types of analog E&M subscriber lines.
89

Examples
# Configure subscriber line 5/0 analog E&M subscriber line type as type 3.
<Sysname> system-view [Sysname] subscriber-line 5/0 [Sysname-subscriber-line5/0] type 3

vi-card busy-tone-detect
Description
Use vi-card busy-tone-detect to configure the parameters for the busy tone detection on the FXO interface. Use undo vi-card busy-tone-detect to restore the default settings. This command applies to the FXO interface only. The system supports four types of busy tones, which are specified by the index argument. When detecting a busy tone on the FXO interface, the system will automatically calculate the parameters related to busy tone detection. You can use the display current-configuration command to display the settings of these parameters. After you use the vi-card busy-tone-detect custom command to configure the parameters related to the busy tone detection, these parameters do not take effect immediately. The manually configured busy tone parameters can take effect only after you execute the area custom command in voice view.

Syntax
vi-card busy-tone-detect { auto index line-number | custom area-number index argu f1 f2 p1 p2 p3 p4 p5 p6 p7 } undo vi-card busy-tone-detect { auto | custom } index

View
Voice view

Default level
2: System level

Parameter
Index: Index of busy tone type, in the range of 0 to 3. line-number: Voice subscriber line number. The value range varies with devices as well as the cards inserted. area-number: Area number. It is set to 2. argu: Reserved, in the range of 0 to 32,767. f1: Frequency 1 in Hz, in the range of 50 to 3,600. f2: Frequency 2 in Hz, in the range of 50 to 3,600. p1: Signal amplitude 1, in the range of 50 to 32,767. p2: Signal amplitude 2, in the range of 50 to 32,767. p3: Duration of a single tone in milliseconds, in the range of 10 to 1,000. p4: Duration error of a single tone in milliseconds, in the range of 0 to 500. p5: Duration of silence in milliseconds, in the range of 10 to 1,000.
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p6: Duration error of silence in milliseconds, in the range of 0 to 500. p7: Absolute difference between p3 and p5 in milliseconds, in the range of 0 to 500

Examples
# Enable the automatic busy tone detection on subscriber line 2/0, with the busy tone index being 0.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] vi-card busy-tone-detect auto 0 2/0

# Manually configure busy tone indexed as 0, duration limit of high/low level, duration error of high/low level, and duration difference of high/low level.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] vi-card busy-tone-detect custom 2 1 99 450 450 8000 8000 800 300 500 500 500

vi-card cptone-custom
Description
Use vi-card cptone-custom to configure parameters for a customized call progress tone. Use undo vi-card cptone-custom to remove the configuration. By default, no customized call progress tone is configured. After you configure parameters for a customized call progress tone, they do not take effect immediately. They do only after you execute cptone country-type CS in voice view.

Syntax
vi-card cptone-custom { busy-tone | congestion-tone | dial-tone | ringback-tone | special-dial-tone | waiting-tone } comb freq1 freq2 time1 time2 time3 time4 undo vi-card cptone-custom { all | busy-tone | congestion-tone | dial-tone | ringback-tone | special-dial-tone | waiting-tone }

View
Voice view

Default level
2: System level

Parameters
busy-tone: Busy tone. congestion-tone: Congestion tone. dial-tone: Dial tone. ringback-tone: Ringback tone. special-dial-tone: Special dial tone. waiting-tone: Call waiting tone. comb: Combination mode, in the range of 0 to 2. The values 0, 1, and 2 represent the superimposition and modulation of two frequencies, and alternation between two frequencies, respectively.

91

freq1 and freq2: Two frequencies in Hz. The frequency range is related to the combination mode. In the case of frequency superimposition or alternation, the two frequencies fall in the range of 300 Hz to 3,400 Hz. In the case of frequency modulation, the two frequencies fall in the range of 300 Hz to 3,400 Hz, and the sum of and the absolute difference between the two frequencies also fall in this range. time1: Make time for the first make-to-break ratio in milliseconds, in the range of 30 to 8,192. In the case of continuous play, the value is 8,192. time2: Break time for the first make-to-break ratio in milliseconds, 30 through 8,191. time3: Make time for the second make-to-break ratio in milliseconds, 30 through 8,191. time4: Break time for the second make-to-break ratio in milliseconds, 30 to 8,191.

Example
# Customize parameters for a busy tone, with the two frequencies both being 425 Hz, and the make time and break time both being 350 milliseconds.
<sysname> system-view [sysname] voice-setup [sysname-voice] vi-card cptone-custom busy-tone 0 425 425 350 350 350 350

vi-card reboot
Description
Use vi-card reboot to reboot a voice card. First use display version or display device to display the distributed slots of the voice cards in the router. Related commands: display version and display device (Fundamentals Command Reference). NOTE: The vi-card reboot command can be used to reboot all analog voice cards (including FXS, FXO, and E&M),
SIC-AUDIO, and BSV.

The SIC digital voice cards and VE1 and VT1 voice cards cannot be rebooted by using commands. You can use the reboot slot slot-number command to reset the analog voice card of FIC. For more information about the reboot slot command, see Fundamentals Command Reference.

Syntax
vi-card reboot slot-number

View
Voice view

Default level
2: System level

Parameters
slot-number: Number of the slot where the voice card is located.

Examples
# Reset the voice card of slot 3.
<Sysname> system-view [Sysname] voice-setup

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[Sysname-voice] vi-card reboot 3

Digital voice subscriber line configuration commands

amd enable
Description
Use amd enable to enable the answering machine detection (AMD) function. Use undo amd enable to disable the AMD function. By default, the AMD function is disabled.

Syntax
amd enable undo amd enable

View
Digital voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable the AMD function on voice subscriber line 1/0:1.
<Sysname> system-view [Sysname] subscriber-line 1/0:1 [Sysname-subscriber-line1/0:1] amd enable

amd parameter
Description
Use amd parameter to configure AMD parameters. Use undo amd parameter to restore the default. By default, the machine-time keyword is 2600 milliseconds; the max-analyze-time keyword is 4000 milliseconds; the min-silence-time keyword is 800 milliseconds; the valid-voice-time keyword is 120 milliseconds; the voice-energy-threshold keyword is 100. There are four AMD detection results: voice, automatic, silence and unknown.

Syntax
amd parameter { machine-time value | max-analyze-time value | min-silence-time value | valid-voice-time value | voice-energy-threshold value } undo amd parameter { machine-time | max-analyze-time | min-silence-time | valid-voice-time | voice-energy-threshold }

93

View
Voice view

Default level
2: System level

Parameters
machine-time value: Sets the answering machine recognition time. If the greeting of the called party lasts longer than the answering machine recognition time, the called party will be considered an answering machine. The value ranges from 10 to 60000 and must be a multiple of 10, in milliseconds. max-analyze-time value: Sets the maximum time for the AMD function to analyze the voice of the speaker. The time starts from the off-hook of the called party. The value ranges from 10 to 60000 and must be a multiple of 10, in milliseconds. min-silence-time value: Sets the minimum silent time after a valid voice. The value ranges from 10 to 60000 and must be a multiple of 10, in milliseconds. valid-voice-time value: Sets the minimum time for the AMD function to detect a valid time. The value ranges from 10 to 60000 and must be a multiple of 10, in milliseconds. voice-energy-threshold value: Sets the voice energy threshold, in the range 10 to 5000.

Examples
# Set the maximum time for the AMD function to analyze the voice of the speaker to 5 seconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] amd parameter max-analyze-time 5000

ani
Description
Use ani to enable the terminating point to request calling party information (service category and calling number) from the originating point during call connection. Use undo ani to disable the terminating point from requesting calling party information from the originating point. By default, the terminating point does not request calling party information from the originating point during call connection. Related commands: cas and ani-offset. NOTE: Configure the local end with this command to support automatic number identification. This command applies to R2 signaling only. Normally the all keyword is configured. Use ka keyword only when required by the connected switch to prevent call
failures.

Syntax
ani { all | ka } undo ani
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View
R2 CAS view

Default level
2: System level

Parameters
all: Specifies the remote end to send the category of the calling party and calling number. ka: Specifies the remote end to send only the category of the calling services.

Examples
# Request the remote office to send calling number category and calling number during call connection.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] ani all

ani-offset
Description
Use ani-offset to configure the number of called number digits that need to be collected prior to requesting calling party information. Use undo ani-offset to restore the default value. Before adequate digits are collected, the system will wait for the next digit until the timer expires. During this period, the system does not request calling party information. It does that only after adequate digits are collected. By default, the number of digits to be collected before receiving calling party information is 1. This command applies to R2 signaling only. Related commands: cas, timer, reverse, and renew. NOTE: Before you can configure this command, you must configure the ani command.

Syntax
ani-offset number undo ani-offset

View
R2 CAS view

Default level
2: System level

Parameters
number: Number of digits to be collected, in the range of 1 to 10.

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Examples
# Start requesting calling number or caller identifier after receiving three digits.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] ani all [Sysname-cas1/0:0] ani-offset 3

answer enable
Description
Use answer enable to configure the originating point to require the terminating point to send answer signal. The two parties begin to talk only after the originating point receives an answer signal. Use undo answer enable to restore the default. By default, the originating party requires the terminating party to send answer signal. This command applies to R2 signaling only. The R2 line signaling coding schemes in some countries do not include answer signal sending. To accommodate to such schemes, you must configure answer enable on the originating point. This allows the terminating point to set up calls after a specified time period. Related commands: re-answer enable and timer dl re-answer.

Syntax
answer enable undo answer enable

View
R2 CAS view

Default level
2: System level

Parameters
None

Examples
# Configure the originating point to disable the terminating point from sending answer signals.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] undo answer enable

callmode
Description
Use callmode to configure the connection mode for an R2 call.
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Use undo callmode to restore the default setting. By default, the connection mode for an R2 call is terminal.

Syntax
callmode { segment | terminal } undo callmode

View
R2 CAS view

Default level
2: System level

Parameters
segment: Specifies the connection mode for an R2 call as segment-to-segment. terminal: Specifies the connection mode for an R2 call as terminal-to-terminal.

Examples
# Set the connection mode for an R2 call to segment.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas 1/0:0] callmode segment

cas
Description
Use cas to enter R2 CAS view, digital E&M signaling view, or digital LGS signaling view. After entering a signaling view, you may configure signaling parameters as desired. When doing that, assign the same value to the ts-set-number keyword in commands cas and timeslot-set. Related commands: timeslot-set, ani-offset, reverse, select-mode, timer, trunk-direction, and renew.

Syntax
cas ts-set-number

View
E1 interface view, T1 interface view

Default level
2: System level

Parameters
ts-set-number: Number of a created timeslot (TS) group, in the range of 0 to 30. The number of a T1 timeslot group ranges from 0 to 23.

Examples
# Enter the R2 CAS view of TS set 5.
<Sysname> system-view

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[Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 5

clear-forward-ack enable
Description
Use clear-forward-ack enable to enable the terminating point to respond with a clear-back signal when the originating point (the calling party) disconnects a call. Use undo clear-forward-ack enable to disable the terminating point from responding with a clear-back signal when the originating point (the calling party) disconnects a call. By default, the terminating point does not send clear-back signals to acknowledge clear-forward signals. This command applies to R2 signaling only. In some countries, if the terminating point controls trunk 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 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. During R2 line signaling exchange, trunk circuit reset is sometimes controlled by the called party (terminating point). The practice in some countries in this case is that after the terminating point receives a clear-forward signal from the originating point, it sends back a clear-back signal as an acknowledgement and then a release-guard signal to indicate that the line at the terminating point side is fully released. Related commands: mode.

Syntax
clear-forward-ack enable undo clear-forward-ack enable

View
R2 CAS view

Default level
2: System level

Parameters
None

Examples
# Enable the terminating point to acknowledge clear-forward signals with clear-back signals.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] clear-forward-ack enable

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display voice subscriber-line

Description
Use display voice subscriber-line command to display subscriber line configuration about voice subscriber line description, echo canceller, echo cancellation sampling time length, comfort noise, and so on.

Syntax
display voice subscriber-line slot-number:{ { ts-set-number | ts-set-number.sub-timeslot } | 15 | 23 } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
slot-number: Voice subscriber line number automatically created upon creation of a TS set or ISDN PRI group. ts-set-number: TS set number. ts-set-number.sub-timelsot: TS set number and TS number. 15: Indicates the subscriber line is created on an E1 interface. 23: Indicates the subscriber line is created on a T1 interface. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the configuration of voice subscriber line 5/0:0.
<Sysname> display voice subscriber-line 5/0:0 Current information Type Status Call Status : TS TS TS TS TS TS TS TS TS 1 2 3 4 5 6 7 8 9 = IDLE = IDLE = IDLE = IDLE = IDLE = IDLE = IDLE = IDLE = IDLE = IDLE ----subscriber-line5/0:0 = R2 = PhysicalDown

TS 10

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TS 11 TS 12 TS 13 TS 14 TS 15 TS 17 TS 18 Description Private Line Cng Echo Canceller Echo Canceller Tail-Length Nlp On Receive Gain Transmit Gain DTMF Threshold Digital PCM Type

= IDLE = IDLE = IDLE = IDLE = IDLE = IDLE = IDLE = subscriber-line5/0:0 Interface = None = Enable = Enable = 32 = Enable = 0.0 = 0.0 = Insensitivty = A-Law

Table 25 Output description Field

Current information Type Status Call Status Description Private Line Cng The subscriber line's description Echo Canceller Echo Canceller Tail-Length Nlp on Receive Gain Transmit Gain DTMF Threshold Digital PCM Type

Description
Information about the current voice subscriber line Signaling type on the voice subscriber line Status of the voice subscriber line Status of the voice protocol call Information about the voice subscriber line Private line dialup mode of the voice subscriber line Comfort noise setting on the voice subscriber line The description of the subscriber line Echo cancellation setting on the voice subscriber line Echo interval setting on the voice subscriber line Setting of nonlinear processing (NLP) in the echo canceller on the voice subscriber line Input gain of the voice subscriber line Output gain of the voice subscriber line DTMF parameters of the digital voice subscriber line Companding law used for signal quantization on the voice subscriber line

dl-bits
Description
Use dl-bits to configure the ABCD bit pattern for R2 signals.
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Use undo dl-bits to restore the defaults. This command applies to R2 signaling only. You may need to use this command to accommodate to the ABCD bit pattern schemes used in different countries. Related commands: seizure-ack enable and answer enable.

Syntax
dl-bits { answer | blocking | clear-back | clear-forward | idle | seize | seizure-ack | release-guard } { received | transmit } ABCD undo dl-bits { answer | blocking | clear-back | clear-forward | idle | seize | seizure-ack | release-guard } { received | transmit }

View
R2 CAS view

Default level
2: System level

Parameters
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. received: Indicates that the signaling setting applies to received R2 line signals. transmit: Indicates that the signaling setting applies to transmitted R2 line signals. ABCD: ABCD bit pattern of R2 line signals, in the range of 0000 to 1 1 1 1. Table 26 Default values of signals in R2 digital line signaling Signal
Answer Blocking Clear-back Clear-forward Idle Seize Seizure-ack Release-guard

Default rx-bits ABCD

0101 1101 1101 1001 1001 0001 1101 1001 101

Default tx-bits ABCD

0101 1101 1101 1001 1001 0001 1101 1001

Examples
# Set the ABCD bit pattern for received R2 idle signal to 1 101, and to 101 for transmitted R2 idle signal. 1
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] dl-bits idle received 1101 [Sysname-cas1/0:0] dl-bits idle transmit 1011

dtmf enable
Description
Use dtmf enable to set the way receiving and transmitting R2 signals to DTMF mode. Use undo dtmf enable to restore the default. By default, MFC mode is adopted. This command applies to R2 signaling only. Related commands: timer dtmf.

Syntax
dtmf enable undo dtmf enable

View
R2 CAS view

Default level
2: System level

Parameters
None

Examples
# Adopt DTMF mode to receive and send R2 signals.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] dtmf enable

dtmf threshold digital

Description
Use dtmf threshold digital to set the DTMF detection sensitivity. Use undo dtmf threshold digital to restore the default DTMF detection sensitivity. By default, the DTMF detection sensitivity level is 0, that is, insensitive.

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The more sensitive the DTMF detection is, the larger the tolerance of DTMF collection is. The possibility of detecting error codes becomes relatively high while the possibility of missing detecting error codes becomes low.

Syntax
dtmf threshold digital value undo dtmf threshold digital

View
BSV voice subscriber line view

Default level
2: System level

Parameters
digital: Sets a digital voice subscriber line. value: 0 or 1. 0 indicates that DTMF detection is insensitive while 1 indicates that DTMF detection is sensitive.

Examples
# Set the DTMF detection to be sensitive.
<Sysname> system-view [Sysname] subscirber-line1/0:0 [Sysname-subscriber-line1/0:0] dtmf threshold digital 1

enable snmp trap updown

Description
Use enable snmp trap updown to enable the interface to generate linkUp/linkDown traps upon link changes. Use undo enable snmp trap updown to disable the interface to generate linkUp/linkDown traps upon link changes. By default, the interface is enabled to generate linkUp/linkDown traps upon link changes.

Syntax
enable snmp trap updown undo enable snmp trap updown

View
BSV BRI interface view

Default level
2: System level

Parameters
None

Examples
# Disable interface BSV BRI 2/0 to generate linkUp/linkDown traps.
<Sysname> system-view [Sysname] interface bri 2/0

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[Sysname-Bri2/0] undo enable snmp trap updown

final-callednum enable
Description
Use final-callednum enable to enable the originating point to send a number terminator to the terminating point after it sends all digits of a called number. After the terminating point receives this terminator, it stops requesting the called number. Use undo final-callednum enable to disable the originating point from sending a number terminator to the terminating point after it sends all digits of a called number. By default, no number terminator is sent. This command applies to R2 signaling only. You may configure final-callednum to accommodate to the R2 interregister signaling in some countries where a number terminator can be sent to indicate that all digits of a called number have been sent. Related commands: register-value digital-end.

Syntax
final-callednum enable undo final-callednum enable

View
R2 CAS view

Default level
2: System level

Parameters
None

Examples
# Enable the originating point to send the number terminator signal.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] final-callednum enable

force-metering enable
Syntax
force-metering enable undo force-metering enable

View
R2 CAS view

Default level
2: System level
104

Parameters
None

Examples
# Enable R2 metering signal processing.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] force-metering enable

group-b enable
Description
Use group-b enable to enable R2 signaling to use Group B signals to complete registers exchange. Use undo group-b enable to disable R2 signaling from using Group B signals to complete registers exchange. By default, Group B signals are used to complete registers exchange. This command applies to R2 signaling only. You may need to configure the undo form of this command to accommodate to the R2 interregister signaling in some countries where Group B signals is not supported or cannot be interpreted correctly. Related commands: register-value req-switch-groupb.

Syntax
group-b enable undo group-b enable

View
R2 CAS view

Default level
2: System level

Parameters
None

Examples
# Adopt Group B signals to complete registers exchange.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] group-b enable

105

line
Description
Use line to configure the binding between a POTS entity and a logical voice subscriber line. Use undo line to remove the binding. By default, there is no binding between a POTS entity and a logical voice subscriber line. After configuring a target match template with match-template for a voice entity, you need to associate the entity with a logical interface to indicate from which interface the traffic destined for the target should be routed. Related commands: entity, pri-set, and timeslot-set.

Syntax
line slot-number:{ ts-set-number | 15 | 23 } undo line

View
POTS voice entity view

Default level
2: System level

Parameters
slot-number: Number of the E1/T1 interface corresponding to a subscriber line. ts-set-number: Number of the TS set created on the E1/T1 interface. 15: Indicates that the POTS voice entity is to be associated with an E1 voice ISDN PRI interface. 23: Indicates that the POTS voice entity is to be associated with a T1 voice ISDN PRI interface.

Examples
# Associate a POTS entity with a TS set on an E1 interface.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] line 1/0:1

mode
Description
Use mode to configure a national R2 signaling variant. Use undo mode to restore the default. By default, ITU-T R2 signaling applies. This command applies to R2 signaling only. The R2 signaling standards implemented in different countries and regions may vary. They are called ITU variants. To accommodate to the R2 signaling in a country or region, you may use the mode command. The

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system can automatically select the appropriate subscriber line state, service category, metering signal, and signal values of C and D bits. At present, the device supports Brazil, Mexico, Argentina, India, New Zealand, Thailand, Bengal, South Korea, Hongkong, Indonesia, and other ITU-T variants. With the default-standard keyword configured, the system initializes the subscriber line status, service type, metering signal and C and D signaling bits and other parameters depending on the default settings of configured national R2 signaling variants. If the custom keyword is configured, you can customize specific signaling exchange procedures and signal values in R2 signaling to accommodate to countries. Related commands: register-value and force-metering enable.

Syntax
mode zone-name [ default-standard ] undo mode

View
R2 CAS view

Default level
2: System level

Parameters
zone-name: Country or region name. The argument can be one of the specified values: argentina: Uses Argentinean R2 signaling standard. australia: Uses Australian R2 signaling standard. bengal: Uses Bengalee R2 signaling standard. brazil: Uses Brazilian R2 signaling standard. china: Uses Chinese R2 signaling standard. custom: Uses customized R2 signaling standard. hongkong: Uses Hongkong R2 signaling standard. india: Uses Indian R2 signaling standard. indonesia: Uses Indonesian R2 signaling standard. itu-t: Uses ITU-T R2 signaling standard. korea: Uses Korean R2 signaling standard. malaysia: Uses Malaysian R2 signaling standard. mexico: Uses Mexican R2 signaling standard. newzealand: Uses New Zealand R2 signaling standard. singapore: Uses Singaporean R2 signaling standard. thailand: Uses Thai R2 signaling standard. default-standard: Initializes R2 signaling parameters such as values of the force-metering command based on national R2 signaling variants.

Examples
# Adopt Hongkong default R2 signaling.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] mode hongkong default-standard

107

pcm
Description
Use pcm to configure a companding law used for quantizing signals. Use undo pcm to restore the default. Companding laws are adopted to quantize signals unevenly for the purpose of reducing noise and improving signal-to-noise ratio. Underpinning this approach are the statistics about voice signals, which indicate that lower power signals are more likely to be present than higher power signals. According to CCITT, when devices in two countries use different companding schemes to communicate, the side using -law is responsible for converting signals to A-law. NOTE: By default, the companding law for VE1 interfaces is A-law, while that for VT1 interfaces is -law.

Syntax
pcm { a-law | -law } undo pcm

View
Voice subscriber line view

Default level
2: System level

Parameters
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. -law: Companding -law, used in North America and Japan.

Examples
# Adopt -law companding for signal quantization.
<Sysname> system-view [Sysname] subscirber-line1/0:0 [Sysname-subscriber-line1/0:0] pcm u-law

posa called-length
Description
Use posa called-length to set the length of called numbers that can be received by the E1POS card. Use undo posa called-length to restore the default. By default, the length of called numbers that can be received by the E1POS card is 31 digits.

Syntax
posa called-length called-length undo posa called-length

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View
R2 CAS view

Default level
2: System level

Parameters
called-length: Sets the length of called numbers that can be received by the E1POS card. The value ranges from 1 to 31 digits.

Examples
# Using R2 signaling to access the POS terminal and set the length of called numbers that can be received by the E1POS card to 8 digits.
<Sysname> system-view [Sysname] controller e1 5/0 [Sysname-e1 5/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 5/0] cas 0 [Sysname-cas5/0:0] posa called-length 8

pri-set
Description
Use pri-set to bundle timeslots on an E1 or T1 interface into a PRI group. Use undo pri-set to remove the bundle. By default, no PRI group is created. When creating a PRI group on a CE1/PRI interface: TS0 is used for FSC, TS16 as a D channel for signaling transmission, and other timeslots as B channels for data transmission. You may bind the timeslots except for timeslot 0 into a PRI group (as the D channel, timeslot 16 is automatically bundled). This PRI group is logically equivalent to an ISDN PRI interface in the form of 30B + D. If no timeslot is specified, all timeslots except for TS0 are bound into an interface similar to an ISDN PRI interface in the form of 30B+D. For the created PRI group, the system automatically creates a serial interface named serial number:15. When creating a PRI group on a T1 interface: TS24 is used as D channel for signaling transmission, and other timeslots as B channels for data transmission. You may randomly bind these timeslots into a PRI group (as the D channel, TS24 is automatically bound). This PRI group is logically equivalent to an ISDN PRI interface in the form of 23B + D. For the created PRI group, the system automatically creates a serial interface named serial number:23.

Syntax
pri-set [ timeslot-list range ] undo pri-set

View
E1 interface view, T1 interface view

Default level
2: System level
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Parameters
range: Specifies timeslots to be bundled. Timeslots are numbered 1 through 31 on an E1 interface and 1 to 24 on a T1 interface. You may specify a single timeslot by specifying a number, a range of timeslots by specifying a range in the form of number1-number2, or several discrete timeslots by specifying number1, number2-number3.

Examples
# On interface E1 1/0 bind timeslots 1, 2, and 8 through 12 into a PRI group.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] pri-set timeslot-list 1,2,8-12

qsig-tunnel enable
Description
Use qsig-tunnel enable to enable the QSIG tunneling function. Use undo qsig-tunnel enable to disable the function. By default, the QSIG tunneling function is disabled.

Syntax
qsig-tunnel enable undo qsig-tunnel enable

View
Digital voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable the QSIG tunneling function.
<Sysname> system-view [Sysname] subscriber-line 1/0:15 [Sysname-subscriber-line1/1:15] qsig-tunnel enable

re-answer enable
Description
Use re-answer enable to enable the originating point to support re-answer signal processing. Use undo re-answer enable to restore the default. By default, the originating point does not support re-answer signal processing. This command applies to R2 signaling 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
110

instead. If it receives the re-answer signal from the terminating point within a specified time, it continues the call; otherwise, it disconnects the call upon timeout. Related commands: answer enable and timer dl re-answer.

Syntax
re-answer enable undo re-answer enable

View
R2 CAS view

Default level
2: System level

Parameters
None

Examples
# Enable the originating point to process re-answer signals.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas 1/0:0] re-answer enable

register-value
Description
Use register-value to configure R2 register signal values. Use undo register-value to restore the defaults. You may set a signal value to 16 to indicate that the signal function does not exist. For example, if the send last digit signal is not available in a national R2 signaling variant, you may set the value for req-lastfirstdigit to 16. The purpose of register-value is to assign values for signals requesting responses from the remote end. For example, after you configure the register-value callingcategory command, the terminating point sends the send calling category signal with the specified value to the originating point for the calling category. This command applies to R2 signaling only. Related commands: group-b enable. NOTE: As some national register signal coding schemes may not support all the register signals mentioned in this section, you are recommended to use defaults unless necessary. For example, the ITU-T recommendation is available with the send calling category signal (the callingcategory keyword) but not the send billing category (billingcategory) signal.

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Syntax
register-value { billingcategory | callcreate-in-groupa | callingcategory | congestion | demand-refused | digit-end | nullnum | req-billingcategory | req-callednum-and-switchgroupa | req-callingcategory | reqcurrentcallednum-in-groupc | req-currentdigit | req- firstcallednum-in-groupc | req-firstcallingnum | req-firstdigit | req-lastfirstdigit | req-lastseconddigit | req-lastthirddigit | req-nextcallednum | req-nextcallingnum | req-switch-groupb | subscriber-abnormal |subscriber-busy | subscriber-charge |subscriber-idle } value undo register-value { billingcategory | callcreate-in-groupa | callingcategory | congestion | demand-refused | digit-end | nullnum | req-billingcategory | req-callednum-and-switchgroupa | req-callingcategory | req- currentcallednum-in-groupc | req-currentdigit | req- firstcallednum-in-groupc | req-firstcallingnum | req-firstdigit | req-lastfirstdigit | req-lastseconddigit | req-lastthirddigit | req-nextcallednum | req-nextcallingnum | req-switch-groupb | subscriber-abnormal |subscriber-busy | subscriber-charge |subscriber-idle }

View
R2 CAS view

Default level
2: System level

Parameters
billingcategory value: Specifies the billing category value, in the range of 1 to 16. It configures the KA signal in R2 signaling. The KA signal is sent by the originating point forward to the originating toll office or originating international exchange to indicate calling category. The signal provides two types of information for this call connection: billing category (regular, immediate, or toll free) and subscriber level (with or without priority). callcreate-in-groupa value: Specifies the direct call setup signal value, in the range of 1 to 16. callingcategory value: Specifies the calling category signal value, in the range of 1 to 16. It configures the R2 KD signal. It functions to identify whether break-in and forced- release can be implemented by or on the calling party. congestion value: Specifies the congestion signal value, in the range of 1 to 16. demand-refused value: Specifies the request-refused signal value, in the range of 1 to 16. digit-end value: Specifies the digit-end signal value, in the range of 1 to 16. nullnum value: Specifies the null number signal value, in the range of 1 to 16. req-billingcategory value: Specifies the send billing category signal value, in the range of 1 to 16. req-callednum-and-switchgroupa value: Specifies the send last digit and changeover to Group A signal value, in the range of 1 to 16. req-callingcategory value: Specifies the send calling category signal value, in the range of 1 to 16. req-currentcallednum-in-groupc value: Specifies the send current called number signal in Group C state, in the range of 1 to 16. req-currentdigit value: Specifies the send current digit signal, in the range of 1 to 16. req-firstcallednum-in-groupc value: Specifies the send first digit signal value in Group C state, in the range of 1 to 16. req-firstcallingnum value: Specifies the send calling number signal value, in the range of 1 to 16.
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req-firstdigit value: Specifies the send first digit signal value, in the range of 1 to 16. req-lastfirstdigit value: Specifies the send last digit signal value, in the range of 1 to 16. req-lastseconddigit value: Specifies the send last second digits signal value, in the range of 1 to 16. req-lastthirddigit value: Specifies the send last three digits signal value, in the range of 1 to 16. req-nextcallednum value: Specifies the send next called number signal value, in the range of 1 to 16. req-nextcallingnum value: Specifies the send next calling number signal value, in the range of 1 to 16. req-switch-groupb value: Specifies the changeover to Group B signal value, in the range of 1 to 16. subscriber-abnormal value: Specifies the subscribers line abnormal signal value, in the range of 1 to 16. subscriber-busy value: Specifies the subscribers line busy signal value, in the range of 1 to 16. subscriber-charge value: Specifies the charge value when the subscribers line is idle, in the range of 1 to 16. subscriber-idle value: Specifies the subscribers line idle value, in the range of 1 to 16. It configures the R2 KB signal used for describing the called subscribers line status, for example, whether the line is idle. It acknowledges and controls call connection. If your router is connected to a PBX, change the KB value on the router to that used on the PBX, in case different KB values are used. If your router is connected to another router, you only need to make sure that the same KB signal value is used between them. The defaults vary by national variant.

Examples
# Request the originating point to send calling category by configuring a backward signal (signal value 7).
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] register-value req-callingcategory 7

renew
Description
Use renew to configure the values of C bit and D bit in R2 signaling. Use undo renew to restore the default. The default value varies with R2 signaling standards in countries. This command applies to R2 signaling only. R2 signaling uses bits A and B to convey real status information while leaving bits C and D constant. The values of bits C and D are national variant dependent. For example, they are fixed to 01 in most countries but 1 in some other countries. 1 You may use this command to adapt values of bits C and D to different line signaling coding schemes. The settings of bits A and B in this command however are not necessarily the real ones during transmission. Related commands: cas and reverse.

Syntax
renew ABCD undo renew

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View
R2 CAS view

Default level
2: System level

Parameters
ABCD: Defines the default of each signal bit in transmission. Each bit can take the value of 0 or 1. The default C and D bit values vary by country mode.

Examples
# Set bits C and D of R2 line signaling to 1 1.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] renew 0011

reverse
Description
Use reverse to configure line signal inversion mode. Use undo reverse to invert ABCD bits of the current line signaling whose values are 1 after reverse is executed. This command applies to R2 signaling only. You may configure an interface to invert the values of any ABCD bits before sending or after receiving a line signal by replacing 0 with 1 or vice versa. Related commands: cas and renew.

Syntax
reverse ABCD undo reverse

View
R2 CAS view

Default level
2: System level

Parameters
ABCD: Indicates whether corresponding ABCD bits in R2 signaling need inversion. Each argument in this command takes either of the two values: 0 for normal or 1 for inversion. The default is 0000, that is, inversion disabled.

Examples
# Invert the values of bits B and D in R2 line signaling.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2

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[Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] reverse 0101

seizure-ack enable
Description
Use seizure-ack enable to configure the originating point to require the terminating point to send seizure acknowledgement signal during R2 line signaling exchange. Use undo seizure-ack enable to restore the default. By default, the originating point requires the terminating point to send seizure acknowledgement signal. This command applies to R2 signaling only. Normally, the terminating point acknowledges received seizure signals. The R2 line signaling coding schemes in some countries however do not require the terminating point to do this. To accommodate these schemes, you can configure the undo seizure-ack enable command, allowing the terminating point not to acknowledge received seizure signals. Related commands: timer dl seizure.

Syntax
seizure-ack enable undo seizure-ack enable

View
R2 CAS view

Default level
2: System level

Parameters
None

Examples
# Disable the terminating point from sending seizure acknowledgement signals.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] undo seizure-ack enable

select-mode
Description
Use select-mode to set the E1 trunk routing mode. Use undo select-mode to restore the default. By default, the timeslot with the smallest number is selected. Related commands: cas and trunk-direction.

115

Syntax
select-mode { max | maxpoll | min | minpoll } undo select-mode

View
R2 signaling view

Default level
2: System level

Parameters
max: Selects the timeslot with the greatest number from currently available timeslots. maxpoll: Selects the timeslot with the greatest number from available timeslots in the first timeslot polling; in later pollings, selects in descending order timeslots with numbers less than the one picked out in the previous polling. Suppose TS31 and TS29 are not available. In the first polling, TS30 will be picked out for use and in the next polling, TS28. min: Selects the timeslot with the lowest number from available timeslots. min: Selects the timeslot with the smallest number from currently available timeslots. minpoll: Selects the timeslot with the lowest number from available timeslots in the first timeslot polling; in later pollings, selects in ascending order timeslots with numbers greater than the one picked out in the previous polling. Suppose TS1 and TS3 are not available. In the first polling, TS2 will be picked out for use and in the next polling, TS4.

Examples
# Set the trunk routing mode for TS set 5 to max on interface E1 1/0.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 5 [Sysname-cas1/0:5] select-mode max

sendring ringbusy enable

Description
Use sendring ringbusy enable to enable the terminating side to send busy tones to calling subscribers. Use undo sendring ringbusy enable to disable the terminating side from sending busy tones to calling subscribers. By default, the terminating point sends busy tones to calling subscribers. This command applies to R2 signaling only. Related commands: timer ring.

Syntax
sendring ringbusy enable undo sendring ringbusy enable

116

View
R2 CAS view

Default level
2: System level

Parameters
None

Examples
# On TS set 5 on interface E1 1/0 configure the terminating point to send ringback tone to the calling side.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 5 [Sysname-cas1/0:5] sendring ringbusy enable

signal-value
Description
Use signal-value to configure the ABCD bit patterns of idle receive, receive seized, idle transmit, and transmit seized signals on the digital E&M voice subscriber line. Use undo signal-value to restore the defaults. By default, the ABCD bit patterns of the receive idle signal and the transmit idle signal are 1 101, and the ABCD bit patterns of the receive seized signal and the transmit seized signal are 0101. After changing the ABCD bit pattern of a digital E&M signal, you must shut down the digital E&M subscriber line with shutdown and then bring the line up with the undo shutdown command. Otherwise, the voice subscriber line cannot work normally. Related commands: subscriber line.

Syntax
signal-value { received idle | received seize | transmit idle | transmit seize } ABCD undo signal-value { received idle | received seize | transmit idle | transmit seize }

View
Digital E&M voice subscriber line view

Default level
2: System level

Parameters
received idle: Indicates the receive idle signal of digital E&M signaling. received seize: Indicates the receive seized signal of digital E&M signaling. transmit idle: Indicates the transmit idle signal of digital E&M signaling. transmit seize: Indicates the transmit seized signal of digital E&M signaling. ABCD: Default ABCD bit pattern during transmission, with each bit taking the value of 0 or 1.

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Examples
# Set the ABCD bit pattern to 101 for the transmit seized signal on digital E&M subscriber line 1/0:0. 1
<Sysname> system-view [Sysname] subscirber-line1/0:0 [Sysname-subscriber-line1/0:0] signal-value transmit seize 1011

special-character
Description
Use special-character to configure the special characters acceptable during register signal exchange. Use undo special-character to remove the configured special characters. By default, no special characters are configured. This command applies to R2 signaling only. You may need to configure this command to accommodate to some national R2 signaling variants where Group I forward signals can represent special characters such as pound signs (#) and asterisks (*) in addition to digits. NOTE: You cannot use special-character to assign a special character different signal values. To make sure that the device can process calls correctly, assign special characters different signal values.

Syntax
special-character character number undo special-character character number

View
R2 CAS view

Default level
2: System level

Parameters
character: Special character, which can be a pound sign (#) or asterisk (*), A, B, C, or D. number: Code of register signal, in the range of 1 to 16. 1

Examples
# Assign the pound sign (#) the register signal code 1 1.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] special-character # 11

118

subscriber-line
Description
Use subscriber-line to enter E1/T1 voice subscriber line view. Upon creation of a TS set on an E1/T1 interface, the system automatically creates a logical voice subscriber line numbered in the form of E1/T1 interface number:TS set number. On the voice subscriber line, you can conveniently configure signaling and other voice functions for the corresponding E1/T1 line. On each E1/T1 interface you can create only one TS set. After you create a PRI group with pri-set on an E1/T1 interface, a voice subscriber line is automatically created. This line is numbered E1 interface-number:15 on an E1 interface and T1 interface-number:23 on a T1 interface. Related commands: timeslot-set and pri-set.

Syntax
subscriber-line slot-number:{ ts-set-number | 15 | 23 }

View
Voice view

Default level
2: System level

Parameters
slot-number: Number of the voice subscriber line automatically created upon creation of a TS set or ISDN PRI group. ts-set-number: Number of the TS set that has been created. 15: Indicates the subscriber line is created for the ISDN PRI group created on an E1 interface. 23: Indicates the subscriber line is created for the ISDN PRI group created on a T1 interface.

Examples
# Enter the view of voice subscriber line 1/0:15.
<Sysname> system-view [Sysname] subscriber-line 1/0:15 [Sysname-subscriber-line1/0:15]

tdm-clock
Description
Use tdm-clock to set the TDM clock source for an E1/T1 interface. Use undo tdm-clock to restore the default. By default, the TDM source clock for an E1POS interface is line TDM clock, and the TDM clock source for other E1 interfaces is the internal clock. When digital voice E1/T1 interfaces perform TDM timeslot interchange, it is important for them to achieve clock synchronization to prevent frame slips and bit errors. Depending on your configurations on E1/T1 interfaces, the system adopts different clocking approaches. When there is a subcard VCPM on the mainboard, the clock distribution principle is:
119

If the line keyword is specified for all interfaces, the clock on the interface with the lowest number is adopted. In case the interface goes down, the clock on the interface with the second lowest number is adopted. If the line primary keywords are specified for one interface, the clock on the interface is adopted. In one system, you can do this on only one interface. If the line keyword is specified for one interface and the internal keyword for all others, the clock on the interface is adopted. Normally, you cannot set the clock source for all interfaces in a system as internal to prevent frame slips and bit errors. You can do this however if the remote E1/T1 interfaces adopt the line clock source.

When there is no VCPM on the mainboard, the configuration of each MIM/FIC is independent but only one interface can be set as line primary.

Syntax
tdm-clock { internal | line [ primary ] } undo tdm-clock

View
E1 interface view, T1 interface view

Default level
2: System level

Parameters
internal: Sets the internal crystal oscillator time division multiplexing (TDM) clock as the TDM clock source on the E1/T1 interface. After that, the E1/T1 interface obtains clock from the crystal oscillator on the mainboard. If it fails to do that, the interface obtains clock from the crystal oscillator on its E1/T1 card. Because SIC cards are not available with crystal oscillator clocks, E1/T1 interfaces on SIC cards can only obtain clock from the mainboard. The internal clock source is also referred to as master clock mode in some features. line: Sets the line TDM clock as the TDM clock source on the E1/T1 interface. After that, the E1/T1 interface obtains clock from the remote device through the line. The line clock source is also referred to as subordinate (slave) clock mode in some features. line primary: Sets the E1/T1 interface to preferably use the line TDM clock as the TDM clock source. After that, the E1/T1 interface always attempts to use the line TDM clock prior to any other clock sources.

Examples
# Set the TDM clock source on interface E1 1/0 to line clock.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] tdm-clock line

timer dl
Description
Use timer dl to configure timeouts of R2 line signals. Use undo timer dl to restore the defaults. This command applies to R2 signaling only.

Syntax
timer dl { answer | clear-back | clear-forward | seizure | re-answer | release-guard } time
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undo timer dl { answer | clear-back | clear-forward | seizure | re-answer | release-guard }

View
R2 CAS view

Default level
2: System level

Parameters
answer time: Timeout time in milliseconds of R2 answer signal, in the range of 100 to 120,000 with a default of 60,000. After the originating point sends a seizure acknowledgement signal, the terminating point should send back an answer signal within the timeout time. If the terminating point fails to send an answer signal within the timeout time, the originating point will clear the connection. Timeout time of R2 answer signal should be configured at both the originating point and the terminating point. The timeout time of answer signals from the terminating point is configured at the originating point, while the timeout time of answer signals for internal function call in a module is configured at the terminating point. clear-back time: Timeout time in milliseconds of R2 clear-back signal, in the range of 100 to 60,000 with a default of .10,000. After the terminating point sends a clear-back signal, it should recognize the forward signal sent back by the originating point within the timeout time. clear-forward time: Timeout time in milliseconds of R2 clear-forward signal configured at the originating point, in the range of 100 to 60,000 with a default of 10,000. After the originating point sends a clear-forward signal, the terminating point should send back a corresponding line signal, clear-back or release guard for example, within the timeout time. seizure time : Timeout time in milliseconds of R2 seizure signal configured at the originating point, in the range of 100 to 5,000 with a default of 1,000. After the originating point sends a seizure signal, the terminating point should send back a seizure acknowledgement signal within the timeout time. re-answer time: Timeout time in milliseconds of R2 re-answer signal configured at the originating point, in the range of 100 to 60,000 milliseconds with a default of 1,000. The originating point releases the line if it does not receive another answer signal from the terminating point after it recognizes the clear-back signal. release-guard time: Timeout time in milliseconds of R2 release guard signal configured at the originating point, in the range of 100 to 60,000 with a default of 1,000. The originating point should send a release guard signal within the timeout time after it receives a clear-back signal from the terminating point in response to a clear-forward signal.

Examples
# Set the timeout time of R2 seizure signal to 300 milliseconds.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] timer dl seize 300

timer dtmf
Description
Use timer dtmf to configure the delay from when the originating point receives a seizure acknowledgement signal to when it starts sending DTMF signals. Use undo timer dtmf to restore the default.
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By default, the delay is 50 milliseconds. This command applies to R2 signaling only. Normally, the originating point starts sending DTMF signals immediately after receiving a line seizure acknowledgement signal. Sometimes, however, you may need to introduce a delay to accommodate to the digit collection process on the remote PBX. Related commands: dtmf enable. NOTE: Before you can configure this command, you must configure the dtmf enable command.

Syntax
timer dtmf time undo timer dtmf

View
R2 CAS view

Default level
2: System level

Parameters
time: Delay before sending a DTMF signal in milliseconds, in the range of 50 to 10,000.

Examples
# Configure the R2 signaling to start sending DTMF signals 800 milliseconds later after receiving a seizure acknowledgement signal.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] dtmf enable [Sysname-cas1/0:0] timer dtmf 800

timer hold
Description
Use timer hold to set the interval for sending keepalive packets. Use undo timer hold to restore the default. By default, the keepalive interval is 10 seconds.

Syntax
timer hold seconds undo timer hold

View
BSV BRI interface view

122

Default level
2: System level

Parameters
seconds: Interval (in seconds) at which the interface sends keepalive packets, in the range 0 to 32767.

Examples
# Set the keepalive interval to 100 seconds for interface BSV BRI 2/0.
<Sysname> system-view [Sysname] interface bri 2/0 [Sysname-Bri2/0] timer hold 100

timer register-pulse persistence

Description
Use timer register-pulse persistence to configure the duration of R2 register pulse signals such as A-3, A-4, and A-6. Use undo timer register-pulse persistence to restore the default, that is, 150 milliseconds. By default, the duration is 150 milliseconds. This command applies to R2 signaling only. When the terminating point sends a backward register pulse signal, A-3 for example, the signal must persist for a specified time period. When the originating point receives the signal, it sends back a Group II forward 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 commands: timer register-complete group-b.

Syntax
timer register-pulse persistence time undo timer register-pulse persistence

View
R2 CAS view

Default level
2: System level

Parameters
persistence time: Duration in milliseconds of R2 register pulse signals, in the range of 50 to 3,000.

Examples
# Set the duration of R2 register pulse signals to 300 milliseconds.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] timer register-pulse persistence 300

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timer register-complete group-b

Description
Use timer register-complete group-b to configure the timeout value of R2 group B signals. After the terminating point switch to Group B, it should send Group B signals within this time period. Use undo timer register-complete group-b to restore the default timeout value of R2 group B signals. By default, the maximum time is 30,000 milliseconds. This command applies to R2 signaling only. Related commands: timer dl.

Syntax
timer register-complete group-b time undo timer register-complete group-b

View
R2 CAS view

Default level
2: System level

Parameters
group-b time: Maximum time in milliseconds that the originating point waits for R2 Group B signals, in the range of 100 to 90,000.

Examples
# Configure the maximum Group B signal exchange time to 10,000 milliseconds.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] timer register-complete group-b 10000

timer ring
Description
Use timer ring to configure the duration of playing a signal tone when R2 signaling is adopted. Use undo timer ring to restore the default duration of playing a signal tone. By default, the duration of playing the ringback tone is 60,000 milliseconds and that of playing the busy tone is 30,000 milliseconds. This command applies to R2 signaling only. Related commands: sendring.

Syntax
timer ring { ringback | ringbusy } time undo timer ring { ringback | ringbusy }

124

View
R2 CAS view

Default level
2: System level

Parameters
ringback time: Sets the duration in milliseconds of playing ringback tone, in the range of 1,000 to 90,000. ringbusy time: Sets the duration in milliseconds of playing busy tone, in the range of 1,000 to 90,000.

Examples
# Set the duration of playing the ringback tone to 10,000 milliseconds when R2 signaling is adopted.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 0 [Sysname-cas1/0:0] timer ring ringback 10000

timeslot-set
Description
Use timeslot-set to create a TS set and specify a signaling mode for it on the E1/T1 interface. Use undo timeslot-set to remove the TS set. By default, no TS set is configured. You can use subscriber-line to enter subscriber line view to configure voice-related attributes only after you create a TS set. Related commands: subscriber-line and cas.

Syntax
timeslot-set ts-set-number timeslot-list timeslots-list signal { e&m-delay | e&m-immediate | e&m-wink | fxo-ground | fxo-loop | fxs-ground | fxs-loop | r2 } undo timeslot-set ts-set-number

View
E1 interface view, T1 interface view

Default level
2: System level

Parameters
ts-set-number: TS set number. For an E1 interface, the TS set number ranges from 0 to 30, and for a T1 interface, the TS set number ranges from 0 to 23. timeslots-list: Timeslot range. Timeslots are numbered 1 through 31 for an E1 interface and 1 through 24 for a T1 interface. TS 16 for an E1 interface (or TS24 for a T1 interface) is used to transmit control signaling. signal: Specifies a signaling mode for the TS set, which should be consistent with that adopted by the central office. It includes certain types of signaling: e&m-delay: Adopts the delay start mode of digital E&M signaling.
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e&m-immediate: Adopts the immediate start mode of digital E&M signaling. e&m-wink: Adopts the wink start mode of digital E&M signaling. fxo-ground: Adopts the FXO ground start mode of digital LGS signaling. fxo-loop: Adopts the FXO loop start mode of digital LGS signaling. fxs-ground: Adopts the FXS ground start mode of digital LGS signaling. fxs-loop: Adopts the FXS loop start mode of digital LGS signaling. r2: Adopts ITU-T Q.421 R2 digital line signaling. This is the one most commonly used.

Examples
# Create TS set 5, including TS1 through TS31 and using R2 signaling.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2

trunk-direction
Description
Use trunk-direction to configure the R2 signal trunking direction. Use undo trunk-direction to restore the default. By default, bidirectional trunking applies. This command applies to R2 signaling only. An incoming trunk carries incoming calls but not outgoing calls while the outgoing trunk does the contrary. A bidirectional trunk carries both incoming calls and outgoing calls. For R2 signaling to operate normally for call connection, you need to make sure that the trunking mode is incoming at one end of the trunk and outgoing at the other end. If both ends are using bidirectional trunking mode, use select-mode to tune trunking policy. This is to prevent timeslot contention. In addition, avoid using bidirectional trunking mode at one end and outgoing mode at the other end, because this can lead to failures of outgoing calls at the end in bidirectional trunking mode. Related commands: cas and select-mode.

Syntax
trunk-direction timeslots timeslots-list { dual | in | out } undo trunk-direction timeslots timeslots-list

View
R2 CAS view

Default level
2: System level

Parameters
timeslots-list: Timeslot range. Timeslots are numbered 1 through 31 on an E1 interface and 1 through 24 on a T1 interface. You may specify a single timeslot by specifying a number, a range of timeslots by specifying a range in the form of number1-number2, or several discrete timeslots by specifying number1, number2-number3. Examples are 1-14, 15, 17-31.
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dual: Bidirectional trunk. in: Incoming trunk. out: Outgoing trunk.

Examples
# Set the trunking mode to bidirectional for TS set 5 on interface E1 1/0.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2 [Sysname-e1 1/0] cas 5 [Sysname-cas1/0:5] trunk-direction timeslots 1-31 dual

ts
Description
Use ts to maintain the trunk circuit of specified timeslots. Related commands: cas. NOTE: The ts query command is available in R2 CAS view, digital E&M CAS view, and digital LGS CAS view.

Syntax
ts { block | open | query | reset } timeslots timeslots-list

View
R2 signaling view

Default level
2: System level

Parameters
block: Blocks the trunk circuit of specified timeslots to make it unavailable. open: Opens the trunk circuit of specified timeslots, allowing it to carry services. query: Queries status of the trunk circuit of specified timeslots to see whether the circuit is busy, open, or blocked in real time. reset: Resets the trunk circuit of specified timeslots when it cannot automatically reset. You may need to do this if the state of an administratively blocked or opened circuit cannot recover for example. timeslots timeslots-list: Specifies a timeslot range. Timeslots are numbered 1 through 31 for an E1 interface and 1 through 24 for a T1 interface. You may specify a single timeslot by specifying a number, a range of timeslots by specifying a range in the form of number1-number2, or several discrete timeslots by specifying number1, number2-number3. Examples are 1-14, 15, 17-31.

Examples
# Reset the circuit of timeslots 1 through 15 in TS5 and query the status of the circuit of TS1 through TS31.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2

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[Sysname-e1 1/0] cas 5 [Sysname-cas1/0:5] ts reset timeslots 1-15 [Sysname-cas1/0:5] ts query timeslots 1-31

128

Dial plan configuration commands

caller-group
Description
Use caller-group to bind a subscriber group to a voice entity. Use undo caller-group to remove the binding of a subscriber group or all subscriber groups to a voice entity. By default, no subscriber group is bound to a voice entity; any calling number is allowed to originate or receive calls. Related commands: subscriber-group.

Syntax
caller-group { deny | permit } subscriber-group-list-number undo caller-group { { deny | permit } subscriber-group-list-number | all }

View
POTS entity view, VoIP entity view, VoFR, interactive voice response (IVR) entity view

Default level
2: System level

Parameters
deny: Refuses calling numbers that match the match templates in a subscriber group to originate or receive calls. permit: Allows calling numbers that match the match templates in a subscriber group to originate or receive calls. subscriber-group-list-number: Subscriber group ID configured by the subscriber-group command, in the range of 1 to 2147483647. all: Specifies all subscriber groups bound to a voice entity.

Examples
# Bind subscriber group 1 to voice entity 1 to allow calling numbers that match subscriber group 1 to originate calls.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1 voip [Sysname-voice-dial-entity1] caller-group permit 1

caller-permit
Description
Use caller-permit to configure a calling number permitted to originate calls.
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Use undo caller-permit to remove the configuration. By default, no calling number is configured, that is, outgoing calls are not restricted. A voice entity allows at most 32 calling numbers to originate calls. Related commands: match-template.

Syntax
caller-permit calling-string undo caller-permit { calling-string | all }

View
POTS entity view, VoIP entity view, VoFR entity view, IVR entity view

Default level
2: System level

Parameters
all: Specifies all calling numbers. calling-string: Calling number permitted to originate a call, in the format of { [ + ] string [ $ ] }| $, with a maximum length of 32 characters. The symbols in the format are: +: Plus sign. The sign itself does not have special meanings. It only indicates that the following string is an effective number and the number is E.164-compliant. $: Dollar sign. When it comes at the end of a number, the calling number must completely match the part before the dollar sign. When it comes alone, the calling number can be null. If there is no sign behind the number, number segments beginning with it are permitted to originate calls. string: A character string consisting of 0123456789#*.!+%[]() -. Table 27 describes these characters.

Table 27 Description of characters in a string Character

0-9 # and * . !

Meaning
Digits 0 through 9. Indicates a valid digit each. Wildcard, which can match any valid digit. For example, 555. can match any number beginning with 555 and ending in four additional characters. Indicates the sub-expression before it appears once or does not appear. For example, 56!1234 can match 51234 and 561234. Indicates the sub-expression before it appears one or more times. However, if a calling number starts with the plus sign, the sign itself does not have special meanings, and only indicates that the following is an effective number and the number is E.164-compliant. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on, and +110022 is an E.164-compliant number. Hyphen (connecting element), used to connect two numbers (The smaller comes before the larger) to indicate a range of numbers, for example, 1-9 inclusive. Indicates the sub-expression before it appears multiple times or does not appear. For example, 9876(54)% can match 9876, 987654, 98765454, 9876545454, and so on.

130

Character
[]

Meaning
Indicates a range for matching. For example, [1-36A] indicates a single character among 1, 2, 3, 6, and A can be matched. Indicates a string of characters. For example, (123) indicates the character string 123. It is usually used together with signs such as !, %, or +. For example, 408(12)+ can match the character string 40812 or 408121212, but not 408 (that is, the string 12 can appear repeatedly and must appear once).

()

NOTE: The sub-expression (one digit or digit string) before signs such as !, %, and + is used for imprecise match. The
processing of these signs is similar to that of the wildcard .. These signs must follow a valid digit or digit string and cannot exist independently.

If embedded, signs [ ] and ( ) must be presented in the form of ( [ ] ). The forms of [ [ ] ] and [ ( ) ] are
incorrect.

The sign -can present itself only in [ ] and characters at the two ends must be of the same type.

Examples
# Configure voice entity 2 to allow the number 660268 to call out.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 2 pots [Sysname-voice-dial-entity2] caller-permit 660268$

# Configure voice entity 2 to allow numbers beginning with 20 to call out.

[Sysname-voice-dial-entity2] caller-permit 20

description
Description
Use description to configure a subscriber group description string. Use undo description to remove the subscriber group description string. By default, no subscriber group description string is configured. The description configured for a subscriber group by using description will not affect the use of the subscriber group. Related commands: match-template and subscriber-group.

Syntax
description text undo description

View
Subscriber group view

Default level
2: System level

131

Parameters
text: Subscriber group description string, consisting of 1 to 80 case-insensitive characters.

Examples
# Identify subscriber group 10 as international.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] subscriber-group 10 [Sysname-voice-dial-group10] description international

dial-prefix
Description
Use dial-prefix to configure a dial prefix for a voice entity. Use undo dial-prefix to remove the configured prefix. By default, no dial prefix is configured. The configuration of the PBX connected to the originating router determines whether a two-stage dialing tone is played or not. When a voice router receives a voice call, it will compare the numbers in the match-templates of its own POTS entities with the received called number and select one POTS entity to process the call. If a prefix is configured, the voice router will send the prefix and dialed number together through the FXO interface. When the number with a prefix exceeds 31 digits, only the first 31 digits are sent. Related commands: match-template and send-number.

Syntax
dial-prefix string undo dial-prefix

View
POTS entity view

Default level
2: System level

Parameters
string: Prefix code, a character string consisting of up to 31 characters that can include 0 through 9, comma, #, and *. Table 28 describes these characters: Table 28 Description of characters in the string argument Character
0-9 , # or *

Meaning
Digits 0 through 9. One comma represents a pause of 500 milliseconds and it can be positioned anywhere in a number. Indicates a valid digit each.

132

Examples
# Specify 0 as a prefix.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 3 pots [Sysname-voice-dial-entity3] dial-prefix 0

display voice subscriber-group

Description
Use display subscriber-group to display the information about a subscriber group or all subscriber groups.

Syntax
display voice subscriber-group { subscriber-group-list-tag | all } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
subscriber-group-list-tag: Specifies a subscriber group ID, which ranges from 1 to 2147483647. all: Specifies all subscriber groups. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the information about the configured subscriber groups.
<Sysname> display voice subscriber-group all Current configuration of subscriber group 1 # Description : <NULL> Referenced by entities: Type: POTS Tag: 2100 Include match templates: Match-template: 1100.. #

133

END Current configuration of subscriber group 2 # Description : <NULL> Referenced by entities: Type: POTS Type: POTS Tag: 2100 Tag: 3100

Include match templates: Match-template: 1200.. # END

Table 29 Output description Field

Current configuration of the appointed subscriber group Description Referenced by entities Type Tag Match-template

Description
Configuration information of a specified subscriber group Description of a subscriber group Information of voice entities that a subscriber group is bound to Type of the voice entity that a subscriber group is bound to Tag of the voice entity that a subscriber group is bound to Match template configured for a subscriber group

display voice number-substitute

Description
Use display voice number-substitute to display the configuration information of a number substitution rule list. Related commands: number-substitute.

Syntax
display voice number-substitute [ list-tag ] [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
list-tag: Serial number of a number substitution rule list, in the range of 1 to 2147483647. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression.
134

include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the configuration information of all number substitution rule lists.
<Sysname> display voice number-substitute Current configuration of number-substitute # ************ NUMBER-SUBSTITUTE ************ List-tag Dot-match rule 1 Input-format Output-format # End : ^011408 : 1408 : 4 : left-right First-rule : INDEX_INVALID

dot-match
Description
Use dot-match to configure the dot match rule of the number substitution rule list. Use undo dot-match to restore the dot match rule to the default. This command only applies to the rules of the number substitution rule list in current view. By default, the dot match rule is end-only. The dots here are virtual match digits. Virtual match digits refer to those matching the variable part such as ., +, %, !, and [] in a regular expression. For example, when 1255 is matched with the regular expression 1[234]55, the virtual match digit is 2, when matched with the regular expression 125+, the virtual match digit is 5, and matched with the regular expression 1..5, the virtual match digits are 25. Related commands: rule.

Syntax
dot-match { end-only | left-right | right-left } undo dot-match

View
Voice number-substitute view

Default level
2: System level

Parameters
end-only: Reserves the digits to which all ending dots (.) in the input number correspond. left-right: Reserves from left to right the digits to which the dots in the input number correspond. right-left: Reserves from right to left the digits to which the dots in the input number correspond.

135

Examples
# Set the dot match rule of number substitution rule list 20 to right-left.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 20 [Sysname-voice-dial-substitute20] dot-match right-left

first-rule
Description
Use first-rule to configure the preferred number substitution rule in the current number substitution rule list. Use undo first-rule to remove the configured preferred number substitution rule. By default, no preferred number substitution rule is configured. In a voice call, the system first uses the rule defined by first-rule for number substitution. If this rule fails to apply or is not configured, it will try to apply all other rules in order until one or none of them applies.

Syntax
first-rule rule-number undo first-rule

View
Voice number-substitute view

Default level
2: System level

Parameters
rule-number: Serial number of a number substitution rule (the serial number of a number substitution rule configured by using the rule command), in the range of 0 to 31.

Examples
# Specify rule 4 in number substitution list 20 as the preferred rule.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 20 [Sysname-voice-dial-substitute20] rule 4 663 3 [Sysname-voice-dial-substitute20] first-rule 4

match-template
Description
Use match-template to configure a calling number match template for a subscriber group. Use undo match-template to delete a calling number match template or all calling number match templates from a subscriber group. By default, no calling number match template is configured for a subscriber group.
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At most 512 calling number match templates can be configured for each subscriber group as long as the total number of calling number match templates for all subscriber groups does not exceed 512. Related commands: description and subscriber-group.

Syntax
match-template match-string undo match-template { match-string | all }

View
Subscriber group view

Default level
2: System level

Parameters
all: Specifies all calling number match templates. match string: Match template in the format of { [ + ] string [ $ ] | $ }, with a maximum length of 31 characters. Each part in a match template can be described thus: +: Plus sign. The sign itself does not have special meanings. It only indicates that the following string is an effective number and the number is E.164-compliant. $: Dollar sign. When it appears at the end of the match template, it indicates the end of a calling number, that is, only the calling number completely matching all characters before $ can originate calls. When it does not appear, calling numbers matching the string argument can originate calls. When it appears separately, it indicates a null calling number. string: A string consisting of any characters of digits 0 through 9, and symbols #, *, ., !, +, %, [, ], (, ), and -.The characters in a string are described in the following table:

Table 30 Meanings of characters in a string Character

0-9 # and * . !

Meaning
Digits from 0 through 9. Each represents a valid digit. Wildcard, which can match any valid digit. For example, 555. can match any number beginning with 555 and ending in four additional characters. Indicates the sub-expression before it appears once or does not appear. For example, 56!1234 can match 51234 and 561234. Indicates the sub-expression before it appears one or more times. However, if a calling number starts with the plus sign, the sign itself does not have special meanings, and only indicates that the following is an effective number and the number is E.164-compliant. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on, and +110022 is an E.164-compliant number. Hyphen (connecting element), used to connect two numbers (the smaller comes before the larger) to indicate a range of numbers, for example, 1-9 inclusive. Indicates the sub-expression before it appears multiple times or does not appear. For example, 9876(54)% can match 9876, 987654, 98765454, 9876545454, and so on. Indicates a range for matching. For example, [1-36] indicates that any character among 1, 2, 3, and 6 can be matched.

% []

137

Character
()

Meaning
Indicates a string of characters. For example, (123) indicates the character string 123. It is usually used together with signs such as !, %, or +. For example, 408(12)+ can match the character string 40812 or 408121212, instead of 408, that is to say, 12 must appear at least once.

NOTE: The sub-expression (one digit or digit string) before signs such as !, %, and + is used for imprecise match. The
processing of these signs is similar to that of the wildcard .. These signs must follow a valid digit or digit string and cannot exist independently.

If embedded, signs [ ] and ( ) must be presented in the form of ( [ ] ). The forms of [ [ ] ] and [ ( ) ] are
incorrect.

The sign - can present itself only in [ ] and characters at the two ends must be of the same type, for example,
0-9. 0-A is not allowed.

Example
# Configure the calling number match template 660268 for subscriber group 2.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] subscriber-group 2 [Sysname-voice-dial-group2] match-template 660268

max-call (voice dial program view)

Description
Use max-call to configure maximum-call-connection sets. Use undo max-call to remove the specified maximum-call-connection set or all maximum-call sets. By default, no maximum-call-connection sets are configured. Together with max-call in voice entity view, this command is used to limit the maximum number of call connections of a voice entity or a set of voice entities. Related commands: max-call (in voice entity view).

Syntax
max-call set-number max-number undo max-call {set-number | all }

View
Voice dial program view

Parameters
set-number: Number identifying a maximum-call-connection set, in the range of 1 to 2,147,483,647. At most 256 maximum-call-connection sets can be configured. max-number: Maximum number of call connections in a maximum-call-connection set, in the range of 0 to 120. all: Specifies all the maximum-call-connection sets.
138

Examples
# Set the maximum number of call connections in maximum-call-connection set 1 to 5.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] max-call 1 5

max-call (voice entity view)

Description
Use max-call to bind a voice entity to the maximum-call-connection set specified by the set-number argument. Use undo max-call to remove the binding. Although you can bind each voice entity to only one maximum-call-connection set, you can change the binding. By default, no maximum-call-connection set is bound, that is, there is no limitation on the number of call connections. Related commands: max-call (in voice dial-program view).

Syntax
max-call set-number undo max-call

View
POTS voice entity view, VoIP voice entity view, VoFR voice entity view, IVR voice entity view

Default level
2: System level

Parameters
set-number: Number identifying a maximum-call-connection set (number of the maximum-call-connection set configured in voice dial program view), in the range of 1 to 2147483647.

Examples
# Bind voice entity 10 to maximum-call-connection set 1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] max-call 1 5 [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] max-call 1

number-match
Description
Use number-match to configure a global number match mode. Use undo number-match to restore the default number match mode. By default, the shortest-number match mode is adopted. Related commands: match-template and terminator.
139

NOTE: If the longest-number match mode is configured and the rule command with the input-format argument ending in a dollar sign ($) is carried out, after a user dials a number, the system will not look up the voice entity to connect the call until the dialing interval expires. Because the dollar sign ($) requires that the last digit configured should match the last one dialed, the system can determine the last dialed digit only after the dialing interval expires and the system stops collecting digits.

Syntax
number-match { longest | shortest } undo number-match

View
Voice dial program view

Default level
2: System level

Parameters
longest: Matches the longest number. shortest: Matches the shortest number.

Examples
# Configure the longest-number match mode.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-match longest

number-priority
Description
Use number-priority peer enable to match a number against a voice entity match template first. Use undo number-priority peer to restore the default. By default, a number starting with * or # will first match against a service feature code.

Syntax
number-priority peer enable undo number-priority peer

View
Voice dial program view

Default level
2: System level

Parameters
None
140

Examples
# Configure a number to first match against a voice entity match template.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-priority peer enable

number-substitute
Description
Use number-substitute to create a number substitution rule list and enter voice number-substitute view. Use undo number-substitute to remove a specified number substitution rule or all number substitution rule lists. By default, no number substitution rule list is configured. Related commands: rule and substitute.

Syntax
number-substitute list-number undo number-substitute { list-number | all }

View
Voice dial program view

Default level
2: System level

Parameters
list-number: Serial number of a number substitution rule list, in the range of 1 to 2147483647. all: Specifies all number substitution rule lists.

Examples
# Enter the voice dial program view and create a number substitution rule list.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 1 [Sysname-voice-dial-substitute1]

priority
Description
Use priority to configure the priority of a voice entity. Use undo priority to restore the priority of a voice entity to the default. By default, the priority level is 0. If you have configured priority levels for voice entities and the selection priority rules (see the select-rule commands), the router will first select the voice entity with the highest priority to initiate a call.
141

Related commands: select-rule.

Syntax
priority priority-order undo priority

View
POTS voice entity view, VoIP voice entity view, VoFR voice entity view, IVR voice entity view

Default level
2: System level

Parameters
priority-order: Priority of a voice entity, in the range of 0 to 10. The smaller the value, the higher the priority.

Examples
# Set the priority level of voice entity 10 to 5.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] priority 5

private-line
Description
Use private-line to configure the PLAR function. Use undo private-line to disable the private line auto ring-down function. This function is disabled by default. This command is applicable to FXO, FXS, analog E&M interface and digital E1/T1 voice interface.

Syntax
private-line string undo private-line

View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice subscriber line view

Default level
2: System level

Parameters
string: E.164 telephone number of the terminating end, a string of 31 digits/characters, which can include 0 through 9, * and #.

Examples
# Configure the private line auto ring-down function on voice subscriber line 1/0 so that 5559262 is automatically dialed out when the subscriber picks up the phone.
<Sysname> system-view

142

[Sysname] subscriber-line1/0 [Sysname-subscriber-line1/0] private-line 5559262

rule
Description
Use rule to configure a number substitution rule. Use undo rule to remove a specified number substitution rule or all number substitution rules. By default, no number substitution rule is configured. After you create a number substitution rule list successfully, you need to use this command to configure specific number substitution rules for it. Related commands: substitute, number-substitute, first-rule, and dot-match.

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 }

View
Voice number-substitute view

Default level
2: System level

Parameters
all: Deletes all number substitution rules. rule-tag: Number identifying a substitution rule, in the range of 0 to 31. input-number: Input string of a number involved in number substitution, in the format of [ ^ ] [ + ] string [ $ ], up to 31 characters. The signs can be explained thus: ^: Caret. The match begins with the first character of a number string. That is, the router begins with the first character of the match string to match a user number. +: Plus sign. The sign itself does not have special meanings. It only indicates that the following string is an effective number and the number is E.164-compliant. $: Dollar sign. It indicates that the last character of the match string must be matched. That is, the last digit of a user number must match with the last character of the match string. string: String consisting of characters such as 0 to 9, #, *, ., !, and %. Table 31 explains these characters:

Table 31 Meanings of characters in the string argument Character

0-9 # and * .

Meaning
Digit 0 through 9. Each indicates a valid digit. Wildcard, which can match any valid digit. For example, 555. can match any number beginning with 555 and ending up with four additional characters.

143

Character
!

Meaning
The character or sub-expression before the sign does not appear or appears only once. For example, 56!1234 can match 51234 and 561234. The character or sub-expression before the plus sign can appear one or more times. However, if a calling number starts with the plus sign, the sign itself does not have special meanings, and only indicates that the following is an effective number and the number is E.164-compliant. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on, and +110022 is an E.164 number. The character or sub-expression before the percent sign does not appear or appears multiple times. For example, 9876(54)% can match 9876, 987654, 98765454, 9876545454, and so on.

output-number: Output string of a number involved in number substitution, in the format of (+)![0-9#*.]+, consisting of up to 31 characters. The characters are described in Table 31. The sub-expression (one digit or digit string) before !, %, or + is not exactly-matched digit(s) and is handled in a similar way the wildcard (.). These signs cannot be used alone and must be preceded by a valid digit or digit string. The dot (.) in the input-number and output-number arguments is handled in these ways:
1.

The dot (.) in the output-number argument is considered invalid. If you use dot-match to set the dot match rule to end-only (that is, only dots at the end of the input number are handled), the dots in the output-number argument are discarded immediately, and the digits which all the dots at the end of the input number correspond to are added to the end of the output number. Extra dots in the output-number argument are discarded. If you use dot-match to set the dot match rule to right-left (from right to left) or left-right (from left to right), and the number of dots in the output-number argument is greater than that in the input-number argument, all digits which the dots in the input-number argument correspond to are selected to replace the dots in the output-number argument one by one from left to right. The remaining dots (that are not replaced) in the output-number argument are discarded. Extra dots in the input-number argument are discarded. If you use dot-match to set the dot match rule to right-left (from right to left) or left-right (from left to right), and the number of dots in the input-number argument is greater than or equal to that in the output-number argument, the dot handling includes two cases: For the right-left dot match rule, digits which the dots in the input-number argument correspond to are extracted from right to left according to the number of dots in the output-number argument to replace the dots in the output-number argument one by one. The digits that are not extracted in the input-number argument are discarded. For the left-right dot match rule, digits which the dots in the output-number argument correspond to are extracted from left to right according to the number of dots in the output-number argument to replace the dots in the output-number argument one by one. The digits that are not extracted in the input-number argument are discarded.

2.

3.

The right-left and left-right dot match rules are only applicable to the dot handling in the input number argument and the extracted digits will always replace the dots in the output-number argument from left to right. number-type: Specifies the type of a number. input-number-type: Type of an input number involved in number substitution. For the values, see Table 32.

144

Table 32 Input number type Number type

abbreviated any international national network reserved subscriber unknown

Description
Abbreviated number Any number International number National number, but not a local network Specific service network number Reserved number Local network number Number of an unknown type

output-number-type: Type of an output number involved in number substitution. For the values, see Table 33. Table 33 Output number type Number type
abbreviated international national network reserved subscriber unknown

Description
Abbreviated number International number National number, but not a local network number Specific service network number Reserved number Local network number Number of an unknown type

numbering-plan: Specifies a numbering plan. input-numbering-plan: Input numbering plan involved number substitution. For the values, see Table 34. Table 34 Input numbering plan Numbering plan
any data isdn national private reserved telex unknown

Description
Any numbering plan Data numbering plan ISDN telephone numbering plan National numbering plan Private numbering plan Reserved numbering plan Telex numbering plan Unknown numbering plan

output-numbering-plan: Numbering plan for an output number involved in number substitution. For the values, see Table 35.
145

Table 35 Output numbering plan Numbering plan

data isdn national private reserved telex unknown

Description
Data numbering plan ISDN telephone numbering plan National numbering plan Private numbering plan Reserved numbering plan Telex numbering plan Unknown numbering plan

Examples
# Configure number substitution rules for number substitution rule list 1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 1

# Configure number substitution rule 1 for number substitution rule list 1 thus: Input number: 91 Output number: 1

[Sysname-voice-dial-substitute1] rule 1 ^91 1

# Configure number substitution rule 2 for number substitution rule list 1 thus: Input number: 92 Output number: 2

[Sysname-voice-dial-substitute1] rule 2 ^92 2

# Configure number substitution rule 3 for number substitution rule list 1 thus: Input number: 93 Output number: 3

[Sysname-voice-dial-substitute1] rule 3 ^93 3

# Configure number substitution rule 3 for number substitution rule list 1 thus: Input number: 93 Output number: 3 Input number type: any Output number type: International Input numbering plan: any Output numbering plan: telex.

[Sysname-voice-dial-substitute1] rule 3 ^93 3 number-type any international numbering-plan any telex

146

select-rule rule-order
Description
Use select-rule rule-order to configure match order of rules for the voice entity selection. Use undo select-rule rule-order to restore the default. By default, the match order of rules for the voice entity selection is exact match->voice entity priority->random selection. You can use select-rule rule-order to configure at most three different rules. The match order of rules determines the application sequence of the rules: If there are multiple rules, the system first selects a voice entity according to the first rule. If the first rule cannot decide which voice entity should be selected, the system applies the second rule. If the second rule still cannot decide a voice entity, the system applies the third rule. If all the rules cannot decide which voice entity should be selected, the system selects a voice entity with the smallest ID.

After the random selection rule is applied, there will be no voice entity selection conflict. Therefore, the random selection rule can only serve as a rule with the lowest priority or serve as a unique rule separately. Related commands: select-rule search-stop, select-rule type-first, and priority.

Syntax
select-rule rule-order 1st-rule [ 2nd-rule [ 3rd-rule ] undo select-rule rule-order

View
Voice dial program view

Default level
2: System level

Parameters
1st-rule: First rule in the match order for voice entity selection. The value ranges from 1 to 4. 2nd-rule: Second rule in the match order for voice entity selection. The value ranges from 1 to 4 but differs from that of 1st-rule. 3rd-rule: Third rule in the match order for voice entity selection. The value ranges from 1 to 4 but differs from those of 1st-rule and 2nd-rule. Table 36 describes the meanings of integers 1 through 4. Table 36 Meanings of integers 1 through 4 Integer
1

Meaning
Exact match

Description
The more digits of a digit string are matched from left to right, the higher the precision is. The system stops using the rule once a digit cannot be matched uniquely. Voice entity priorities are divided into 11 levels numbered from 0 to 10. The smaller the value is, the higher the priority is. That means level 0 has the highest priority.

Priority

147

Integer
3 4

Meaning
Random selection Longest idle time

Description
The system selects at random a voice entity from a set of qualified voice entities. The longer the voice entity is idle, the higher the priority is.

Examples
# Set the match order of rules for the voice entity selection is exact match->priority->longest idle time.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] select-rule rule-order 1 2 4

select-rule search-stop
Description
Use select-rule search-stop to configure the maximum number of voice entities found before a search process stops. Use undo select-rule search-stop to restore the default. By default, the maximum number of voice entities found before a search process stops is 128. The select-rule search-stop command is used to define the maximum number of qualified voice entities to be found before a search process stops. Even if the number of voice entities meeting call requirements is greater than max-number, the system will make call attempts to only the maximum number (max-number) of voice entities that are matched in accordance with rules. Related commands: select-rule rule-order and select-rule type-first.

Syntax
select-rule search-stop max-number undo select-rule search-stop

View
Voice dial program view

Default level
2: System level

Parameters
max-number: Maximum number of voice entities found before a search process stops, in the range 1 to 128.

Examples
# Configure the maximum number of voice entities found to 5.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] select-rule search-stop 5

148

select-rule type-first
Description
Use select-rule type-first to configure a rule for voice entity type selection priority. Use undo select-rule type-first to remove a rule for voice entity type selection priority. By default, voice entities are not selected by type. The command is used to configure the sequence of voice entity type selection priority. If different types of voice entities are qualified for a call connection, the system selects a suitable voice entity according to the voice entity type selection priority rule configured by the select-rule type-first command. The order of inputting the parameters determines voice entity type priorities. The system selects the first type first, then the second type, and the third type, finally the fourth type. Related commands: select-rule rule-order and select-rule search-stop.

Syntax
select-rule type-first 1st-type 2nd-type 3rd-type [ 4th-type ] undo select-rule type-first

View
Voice dial program view

Default level
2: System level

Parameters
1st-type: Serial number of the type of the first priority, in the range of 1 to 4. Table 37 describes these values: 2nd-type: Serial number of the type of the second priority, in the range of 1 to 4. The value must be different from that of 1st-type. 3rd-type: Serial number of the type of the third priority, in the range of 1 to 4. The value must be different from that of 1st-type and 2nd-type. 4th-type: Serial number of the type of the fourth priority, in the range of 1 to 4. The value must be different from that of 1st-type, 2nd-type, and 3rd-type. Table 37 describes the meanings of these values. Table 37 Meanings of values Value
1 2 3 4

Meaning
POTS voice entity VoIP voice entity VoFR voice entity IVR voice entity

Examples
# Configure the system to select voice entities in the order of VoIP->POTS->VoFR->IVR.
<Sysname> system-view [Sysname] voice-setup

149

[Sysname-voice] dial-program [Sysname-voice-dial] select-rule type-first 2 1 3 4

select-stop
Description
Use select-stop to disable the voice entity search function. Use undo select-stop to enable the voice entity search function. By default, the voice entity search function is enabled. Related commands: select-rule rule-order, select-rule type-first, and select-rule search-stop.

Syntax
select-stop undo select-stop

View
POTS voice entity view, VoIP voice entity view, VoFR voice entity view, IVR voice entity view

Default level
2: System level

Parameters
None

Examples
# Disable the voice entity search function for voice entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] select-stop

send-number
Description
Use send-number to configure the number sending mode. Use undo send-number to restore the default number sending mode. By default, the truncate mode is used. This command applies to only POTS voice entities. This command is used to control how to send called numbers to PSTN. You can specify to send some digits (defined by the digit-number argument from right to left) or all digits of called numbers. You can also specify to send truncated called numbers, the ending digits of called numbers that match the dot (.). The dot represents the digits that match the variable part in a regular expression. For more information, see Voice Configuration Guide. Related commands: dot-match and match-template.

Syntax
send-number { digit-number | all | truncate }
150

undo send-number

View
POTS entity view

Default level
2: System level

Parameters
digit-number: Number of digits (that are extracted from the end of a number) to be sent, in the range of 0 to 31. It is not greater than the total number of digits of the called number. all: Sends all digits of a called number. truncate: Sends a truncated called number.

Examples
# Configure voice entity 10 to send the last six digits of a called number.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] send-number 6

subscriber-group
Description
Use subscriber-group to create a subscriber group and enter subscriber group view, or directly enter the subscriber group view if the subscriber group already exists. Use undo subscriber-group to delete a subscriber group or all subscriber groups. By default, no subscriber group is created. At most ten subscriber groups can be configured for the system. Related commands: description and match-template.

Syntax
subscriber-group list-number undo subscriber-group { list-number | all }

View
Voice dial program view

Default level
2: System level

Parameters
list-number: Subscriber group ID, in the range of 1 to 2147483647. all: Specifies all subscriber groups.

Examples
# Enter voice dial program view and create a subscriber group.
151

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] subscriber-group 1 [Sysname-voice-dial-group1]

substitute (voice subscriber line view, voice entity view)

Description
Use substitute to bind a calling/called number substitution rule list to the voice subscriber line or voice entity. Use undo substitute to remove the binding between a calling/called number substitution rule list and the voice subscriber line or voice entity. By default, no number substitution rule list is bound to a voice subscriber line or voice entity. That is to say, no number substitution is performed. Before carrying out the this command, you must first use the number-substitute list-number command to configure a number substitution rule list in voice dial program view, and then use rule to configure rules for the list. According to network requirements, you can complete number substitution in the following two ways: Before a voice entity is matched, you can use substitute in subscriber line view to substitute the calling/called number specific to a subscriber line. After a voice entity is matched but before a call is initiated, you can use substitute in voice entity view to substitute a specified calling/called number.

Related commands: number-substitute and rule.

Syntax
substitute { called | calling } list-number undo substitute { called | calling }

View
POTS voice view, VoIP voice view, VoFR voice view, subscriber line voice entity view

Default level
2: System level

Parameters
called: Applies the number substitution rule to a called number. calling: Applies the number substitution rule to a calling number. list-number: Serial number of a number substitution rule list configured by using the number-substitute command), in the range of 1 to 2147483647.

Examples
# Apply number substitution rule list 6 to the called number of the voice subscriber line 1/0.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] subscriber-line1/0 [Sysname-voice-line1/0] substitute called 6

152

substitute (voice dial program view)

Description
Use substitute to bind the calling/called number of incoming/outgoing calls to the specified number substitution rule list. Use undo substitute to remove the binding. By default, no number substitution rule list is bound. That is to say, no number substitution is performed. You should follow these rules when using this command: At most 32 number substitution rule lists can be bound. The system does not stop searching the bound number substitution rule lists in sequence until one rule is applied successfully.

Related commands: number-substitute and rule. NOTE: Outgoing and incoming calls are relative to the IP network. Calls to the IP network are incoming calls, and calls from the IP network or PSTN to PSTN are outgoing calls.

Syntax
substitute { incoming-call | outgoing-call } { called | calling } list-number undo substitute { incoming-call | outgoing-call } { called | calling } { list-number | all }

View
Voice dial program view

Default level
2: System level

Parameters
incoming-call: Binds the calling/called number of incoming calls to the number substitution rule list. outgoing-call: Binds the calling/called number of outgoing calls to the number substitution rule list. called: Applies the number substitution rule to a called number. calling: Applies the number substitution rule to a calling number. all: Specifies all number substitution rule lists. list-number: Serial number of a number substitution rule list configured by using the number-substitute command), in the range of 1 to 2147483647.

Examples
# Apply number substitution rule list 5 to called numbers of incoming calls.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] substitute incoming-call called 5

# Apply number substitution rule lists 5, 6, and 8 to called numbers of outgoing calls.
<Sysname> system-view

153

[Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] substitute outgoing-call called 5 [Sysname-voice-dial] substitute outgoing-call called 6 [Sysname-voice-dial] substitute outgoing-call called 8

terminator
Description
Use terminator to configure a special character as the dial terminator for length-variable telephone numbers. Use undo terminator to remove the dial terminator configuration. By default, no dial terminator is configured. If you set the argument character to # or *, and if the first character of the configured entity number is the same as the argument character (# or *), the device will take this first character as a common number rather than a dial terminator. Related commands: match-template and timer.

Syntax
terminator character undo terminator

View
Voice dial program view

Default level
2: System level

Parameters
character: Dial terminator, which can be any of 0 through 9, pound sign (#), or asterisk (*).

Examples
# Specify the pound sign (#) as the dial terminator.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] terminator #

154

SIP configuration commands

address sip
Description
Use address sip to configure SIP routing for the VoIP voice entity. Use undo address sip to remove specified SIP routing configuration. By default, no routing policy is configured for the VoIP voice entity. Related commands: address sip server-group.

Syntax
address sip { dns domain-name [ port port-number ] | enum-group group-number | ip ip-address [ port port-number ] | proxy | server-group index } undo address sip { dns | ip | proxy }

View
VoIP voice entity view

Default level
2: System level

Parameters
dns domain-name: Domain name of the called entity, which consists of character strings separated by a dot (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.), with a maximum length of 255 characters. enum-group group-number: Number of an ENUM translation rule group, ranging from 1 to 15. port port-number: Port number of the address corresponding to the domain name, in the range of 1 to 65535. ip ip-address: IP address of the peer VoIP gateway. port port-number: Port number, in the range of 1 to 65535. proxy: Uses the SIP proxy server to route outbound calls.

Examples
# Configure the IP address of the peer VoIP gateway as 3.3.3.3 for voice entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] address sip ip 3.3.3.3

# Configure the domain name of the called entity as cc.news.com for voice entity 10.
<Sysname> system-view [Sysname] voice-setup

155

[Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] address sip dns cc.news.com

call-fallback
Description
Use call-fallback register to enable re-registration in the case of a call failure. Use undo call-fallback register to disable call failure-triggered re-registration. By default, call failure-triggered re-registration is disabled.

Syntax
call-fallback register undo call-fallback register

View
SIP client view

Default level
2: System level

Parameters
None

Examples
# Enable all failure-triggered re-registration.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] call-fallback register

crypto
Description
Use crypto to reference an SSL server/client policy when TLS is used as the transport layer protocol for SIP sessions. Use undo crypto to remove the configuration. By default, no SSL policy is referenced. The SSL policies to be referenced must have been configured. You need to first configure the TLS server and client policies, and then specify TLS as the transport layer protocol for incoming SIP calls through the listen transport command; otherwise, no TLS requests can be received. If the TLS server policy or its name is modified, you need to specify TLS as the transport layer protocol again through the transport command, and then the new policy will take effect. If the TLS client policy or its name is modified, the new configuration will take effect for new TLS connections and the current TLS connections still use the original policy.
156

Related commands: listen transport.

Syntax
crypto { ssl-server-policy server-policy-name | ssl-client-policy client-policy-name } undo crypto { server-policy | client-policy }

View
SIP client view

Default level
2: System level

Parameters
ssl-server-policy server-policy-name: References an SSL server policy. The policy name is a string of 1 to 16 case-insensitive characters. ssl-client-policy client-policy-name: References an SSL client policy. The policy name is a string of 1 to 16 case-insensitive characters.

Examples
# Reference SSL server policy Server1 and SSL client policy Server2.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] crypto ssl-server-policy Server1 [Sysname-voice-sip] crypto ssl-client-policy Server2

display voice sip call-statistics

Description
Use display voice sip call-statistics to display the statistics about all SIP calls.

Syntax
display voice sip call-statistics [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

157

Examples
# Display the statistics about all SIP calls.
<Sysname> display voice sip call-statistics Message Statistics of Stack: TPT Message UDP TCP SCTP TLS Total

---------------------------------------------------------------InMsg OutMsgSucc OutMsgFail TXN Message 44 33 0 0 0 0 0 0 0 0 0 0 44 33 0

Inv_Cli NonInv_Cli Inv_Srv NonInv_Srv

---------------------------------------------------------------Create Succ Create Fail Terminal Abnom Request Message In: Out: Response Message In: Out: Error Statistics: --------------------------------------callCb creation failures: call-leg creation failures: transaction creation failures: callCb locate failures: call-leg locate failures: transaction locate failures: user not registered: user not available: request with missing headers: response-no To tag in response: response - invalid via: messages without headers rcvd: SDP decode failures: registration timeouts: retransmitted requests received: transaction timeouts: 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 Inv 0 10 Ack 0 10 10 0 0 Bye 1 4 Can 0 3 1xx 21 0 12 0 0 Opt 0 0 2xx 13 1 Reg 0 5 3xx 0 0 0 0 0 Inf 0 0 4xx 9 0 1 0 0 Prk Upd 0 0 5xx 0 0 0 0 6xx 0 0

----------------------------------------------------------------

----------------------------------------------------------------

158

Table 38 Output description Field

TPT Message

Description
Statistics about SIP transport layer messages, including UDP, TCP, SCTP, and TLS. The messages of each type fall into InMsg, (received), OutMsgSucc (transmitted successfully), and OutMsgFail (sending failure) Statistics of SIP transaction messages. These messages fall into: Inv_Cli (INVITE transaction of client) NonInv_Cli (Non-INVITE transaction of client) Inv_Srv (INVITE transaction of server)

TXN Message

NonInv_Srv (Non-INVITE transaction of server) Each type of message can be displayed by: Create Succ (Creation success) Create Fail (Creation failure) Terminal Abnom (Terminal exception) Statistics of all SIP request messages, including Inv (INVITE), ACK , BUE, Can (CANCEL), Opt (OPTIONS), Reg (GEGISTER), Inf (Information), Prk (PRACK), Upd (UPDATE)

Request Message

Each type of message can be displayed by: In (received) Out (sent) Statistics of all SIP response messages, including 1XX, 2XX, 3XX, 4XX (Cancel), 5XX, and 6XX

Response Message

Each type of message can be displayed by: In (received) Out (sent)

callCb creation failures call-leg creation failures transaction creation failures callCb locate failures call-leg locate failures transaction locate failures user not registered user not available request with missing headers response-no To tag in response response - invalid via messages without headers rcvd SDP decode failures registration timeouts retransmitted requests received

Statistics of call control block creation failures in SIP Statistics of call leg creation failures in SIP Statistics of transaction creation failures in SIP Statistics of call control block location failures in SIP Statistics of call leg location failures in SIP Statistics of transaction location failures in SIP Statistics of user not registered message in SIP Statistics of user not available message in SIP Statistics of request messages with missing headers in SIP Statistics of response messages without the To Tag field in SIP Statistics of response messages with an invalid via field in SIP Statistics of received messages without headers in SIP Statistics of SDP decoding failures in SIP Statistics of registration timeouts in SIP Statistics of received retransmission requests in SIP 159

Field
transaction timeouts

Description
Statistics of transaction timeouts in SIP

display voice sip connection

Description
Use display voice sip connection to display information about SIP connections over a specific transport layer protocol, including both established and attempted connections.

Syntax
display voice sip connection { tcp | tls } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

Parameters
tcp: Displays the information of all TCP connections. tls: Displays the information of all TLS connections. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the information about SIP connections over TCP.
<Sysname> display voice sip connection tcp Conn-Id 569 570 571 572 Local-IP 100.1.1.84 100.1.1.84 100.1.1.84 192.168.0.82 Local-Port 1593 1594 1595 1596 Remote-IP 100.1.1.100 100.1.1.101 100.1.1.81 192.168.0.81 Remote-Port 5060 5060 5060 5060 Conn-State Established Established Established Established +------------------------------------------------------------------------------+

# Display the information about SIP connections over TLS.

<Sysname> display voice sip connection tls Conn-Id 73 Local-IP 192.168.0.202 Local-Port 1086 Remote-IP 192.168.0.132 Remote-Port 5061 Conn-State Established +------------------------------------------------------------------------------+

160

display voice enum-group

Description
Use display voice enum-group to display the configuration information of ENUM translation rule groups.

Syntax
display voice enum-group { all | mark group-number } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
all: Displays all ENUM translation rule groups. mark group-number: Displays the specified ENUM translation rule group with a number from 1 to 15. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display all ENUM translation rule groups.
<Sysname> display voice enum-group all Current configuration of ENUM groups # enum-group 1 rule 1 preference 1 408...(8333) 555\1 cc.news.com # enum-group 2 rule 2 preference 3 408...(8333) 888\1 cc.news2.com # End

Table 39 Output description Field

Current configuration of ENUM groups enum-group 1 rule 1 preference 1 408(8333) 5555\1 cc.news.com

Description
Configuration information of ENUM translation rule groups. ENUM translation rule group. ENUM translation rule.

161

display voice sip dns-record

Description
Use display voice sip dns-record to display SIP DNS records.

Syntax
display voice sip dns-record [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
dns record: Displays DNS records for SIP. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display SIP DNS records.
<Sysname> display voice sip dns-record No. 1 2 Host sip1.hp.com sip2.8056.com IP 100.1.1.163:5060 100.1.1.16:5060

Table 40 Output description Field

No. Host IP

Description
Sequence number of the DNS record Domain name IP address of the domain name

display voice sip reason-mapping

Description
Use display voice sip reason-mapping pstn-sip to query the PSTN release cause code to SIP status code mappings. Use display voice sip reason-mapping sip-pstn to query the SIP status code to PSTN release cause code mappings.

Syntax
display voice sip reason-mapping { pstn-sip | sip-pstn } [ | { begin | exclude | include } regular-expression ]
162

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Query the PSTN release cause code to SIP status code mappings. For the convenience of query, the user-defined SIP status codes are highlighted with an asterisk.
<Sysname> display voice sip reason-mapping pstn-sip Release reason mapping of PSTN to SIP: Index 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 PSTN-Reason 1 2 3 16 17 18 19 20 21 22 23 25 26 27 28 29 31 34 38 41 42 47 55 57 SIP-Status 400* 404 404 --486 408 480 480 403 410 410 500 404 502 484 501 480 503 503 503 503 503 403 403 Default 404 404 404 --486 408 480 480 403 410 410 500 404 502 484 501 480 503 503 503 503 503 403 403

------------------------------------------------------

163

25 26 27 28 29 30 31 32 33 34

58 63 65 70 79 87 88 102 111 127

503 500 488 488 501 403 503 504 500 500

503 500 488 488 501 403 503 504 500 500

Table 41 Output description Field

PSTN-Reason SIP-Status

Description
PSTN release cause code SIP status code corresponding to a PSTN release cause code (If the configured SIP status code is different from the default, it is highlighted with an asterisk.) Default SIP status code corresponding to a PSTN release cause code

Default

# Query the SIP status code to PSTN release cause code mappings. For the convenience of query, the user-defined PSTN release cause codes are highlighted with an asterisk.
<Sysname> display voice sip reason-mapping sip-pstn Release reason mapping of SIP to PSTN: Index 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 SIP-Status 400 401 402 403 404 405 406 407 408 410 413 414 415 416 420 421 423 480 481 PSTN-Reason 127* 21 21 21 1 63 79 21 102 22 127 127 79 127 127 127 127 18 41 Default 41 21 21 21 1 63 79 21 102 22 127 127 79 127 127 127 127 18 41

------------------------------------------------------

164

20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37

482 483 484 485 486 487 488 500 501 502 503 504 505 513 600 603 604 606

25 25 28 1 17 127 127 41 79 38 41 102 127 127 17 21 1 58

25 25 28 1 17 127 127 41 79 38 41 102 127 127 17 21 1 58

Table 42 Output description Field

SIP-Status PSTN-Reason

Description
SIP status code PSTN release cause code corresponding to a SIP status code (If the configured PSTN release cause code is different from the default, it is highlighted with an asterisk.) Default PSTN release cause code corresponding to a SIP status code

Default

dns-type
Description
Use dns-type to set the DNS lookup mode. Use undo dns-type to restore the default mode. The default DNS lookup mode is a-record. If you configure the destination port in the address sip { dns domain-name [ port port-number ] | enum-group group-number }, proxy dns domain-name [ port port-number ], or mwi-server dns domain-name [ port port-number ] command, the DNS lookup mode can only be Type-A. Related commands: address sip, proxy, and mwi-server.

Syntax
dns-type { a-record | srv } undo dns-type

View
SIP client view
165

Default level
2: System level

Parameters
a-record: Sets the DNS lookup mode to Type-A. srv: Sets the DNS lookup mode to SRV.

Examples
# Set the DNS lookup mode to SRV.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] dns-type srv

display voice sip register-state

Description
Use display voice sip register-state to display status information of all user numbers to be registered on the SIP UA.

Syntax
display voice sip register-state [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display all registration status information on the SIP UA.
<Sysname> display voice sip register-state Number 105 2000 3000 Entity 105 107 109 Registrar Server 100.1.1.1:5060 100.1.1.1:5060 cc.news.com:1120 Expires Status N/A 200 N/A login online login +-----------------------------------------------------------------------+

166

Table 43 Output description Field

Number Entity Registrar Server Expires

Description
User number Entity number Address of the registrar, in the format of IP address + port number or domain name + port number Aging time for a user number in seconds State in which a number stays, including:

Status

offline online login logout dnsin: DNS query is being performed before the number is registered dnsout: DNS query is being performed before the number is deregistered

early-media enable
Description
Use early-media enable to enable early media negotiation on the device. When the device is the called party, it sends a 183 session progress response with media information to the calling party. Use undo early-media enable to disable early media negotiation on the device. In other words, when the device is the called party, it sends a 183 ring response without media information to the calling party. By default, the early media negotiation is enabled on the device. When the device is the called party, it sends a 183 session progress response with media information to the calling party.

Syntax
early-media enable undo early-media enable

View
SIP client view

Default level
2: System level

Parameters
None

Examples
# Disable early media negotiation on the device.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] undo early-media enable

167

enum-group
Description
Use enum-group to create an ENUM translation rule group. Use undo enum-group to delete an ENUM translation rule group. By default, no ENUM translation rule group exists.

Syntax
enum-group group-number undo enum-group { group-number | all }

View
Voice dial program view

Default level
2: System level

Parameters
group-number: Number of the ENUM translation group, ranging from 1 to 15. all: Deletes all ENUM translation rule groups.

Examples
# Create ENUM translation rule group 1 and enter the ENUM translation view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] enum-group 1 [Sysname-voice-dial-enumgroup-1]

keepalive
Description
Use keepalive to set the keepalive mode. Use undo keepalive to restore the default keepalive mode. Related commands: redundancy mode.

Syntax
keepalive { options [ interval seconds ] | register } undo keepalive

View
SIP client view

Default level
2: System level

Parameters
options: Sets the keepalive mode to options.
168

interval seconds: Sets the interval for sending options packets in seconds. It ranges from 5 to 65535 and defaults to 60. register: Sets the keepalive mode to register.

Examples
# Set the keepalive mode to options and set the interval for sending options packets to 30 seconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] keepalive options interval 30

line-check enable
Description
Use line-check enable to enable checking the status of voice subscriber lines associated with POTS voice entities. Use undo line-check to disable checking the status of voice subscriber lines associated with POTS voice entities. By default, before registering numbers for a POTS voice entity, the device checks the status of the voice subscriber line associated to the POTS voice entity. The device can send REGISTER requests for numbers only when the status of the line is up. Related commands: line and shutdown (voice subscriber line view).

Syntax
line-check enable undo line-check

View
SIP client view

Default level
2: System level

Parameters
None

Examples
# Disable checking the status of voice subscriber lines associated with POTS voice entities. In other words, as long as a POTS subscriber line is configured, the device can send REGISTER requests for numbers even if the voice subscriber line is shut down.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] undo line-check

169

listen transport
Description
Use listen transport to enable the listening port for the transport layer protocol. Use undo listen transport to restore the default. By default, both the UDP and TCP listening ports are enabled, and the TLS listening port is disabled. You can execute this command multiple times to specify multiple transport layer protocols for incoming SIP calls, and the configured transport layer protocols do not affect one another. Execute listen transport in either of the following scenarios: If the device is the call receiver, you need to enable the listening port of the transport layer protocol used by the incoming calls. The transport layer protocol specified in the registrar command must have been specified with the listen transport command; otherwise, no register request can be initiated.

You need to first configure the TLS server and client policies, and then specify TLS as the transport layer protocol for incoming SIP calls through the listen transport command; otherwise, the execution of listen transport tls will not take effect. If the TLS or TCP is specified as the transport layer protocol, the execution of undo listen transport deletes the established connections. Related commands: registrar and transport.

Syntax
listen transport { tcp | tls | udp } undo listen transport { tcp | tls | udp }

View
SIP client view

Default level
2: System level

Parameters
udp: Specifies UDP as the transport layer protocol for incoming SIP calls and enables UDP listening port 5060. tcp: Specifies TCP as the transport layer protocol for incoming SIP calls and enables TCP listening port 5060. tls: Specifies TLS as the transport layer protocol for incoming SIP calls and enables TLS listening port 5061.

Examples
# Specify TLS as the transport layer protocol for incoming SIP calls.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] listen transport tls

170

media-protocol
Description
Use media-protocol to specify the media flow protocol(s) for SIP calls. Use undo media-protocol to restore the default. By default, SIP calls use RTP as the media flow protocol. When both the RTP and SRTP protocols are specified as the media flow protocols for SIP calls: If the device is the call initiator, both two media flow protocols are carried in the INVITE message for the receiver to select. If the device is the call receiver, the SRTP protocol is first used for media flow negotiation. If the negotiation fails, the RTP protocol is used.

Syntax
media-protocol { rtp | srtp } * undo media-protocol

View
SIP client view

Default level
2: System level

Parameters
rtp: Specifies the RTP as the media flow protocol for SIP calls. srtp: Specifies the SRTP as the media flow protocol for SIP calls.

Examples
# Specify SRTP as the media flow protocol for SIP calls.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] media-protocol srtp

outband sip
Description
Use outband sip to configure the out-of-band SIP DTMF transmission mode. Use undo outband sip to restore the default DTMF transmission mode. By default, the inband DTMF transmission mode is adopted.

Syntax
outband sip undo outband

View
POTS entity view, VoIP entity view
171

Default level
2: System level

Parameters
None

Examples
# Configure the out-of-band SIP DTMF transmission for VoIP entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] address sip ip 10.1.1.2 [Sysname-voice-dial-entity10] outband sip

outbound-proxy
Description
Use outbound-proxy to configure the outbound proxy server information for the SIP UA. Use undo outbound-proxy to remove the outbound proxy server information for the SIP UA. By default, no outbound proxy server information is configured for a SIP UA. For more information about DTMF H.225 out-of-band transmission, DTMF H.245 out-of-band transmission, and DTMF named NTE transmission, see Voice Configuration Guide.

Syntax
outbound-proxy { dns domain-name | ipv4 ip-address } [ port port-number ] undo outbound-proxy { dns | ipv4 }

View
SIP client view

Default level
2: System level

Parameters
dns domain-name: Domain name of the outbound proxy server, which consists of character strings separated by a dot, for example, aabbcc.com. Each separated string contains no more than 63 characters. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.), with a maximum length of 255 characters. ipv4 ip-address: IPv4 address of the outbound proxy server. port port-number: Port number of the outbound proxy server, in the range 1 to 65535.

Examples
# Configure IP address 169.54.5.10 and port number 1 120 of the outbound proxy server for the SIP UA.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] outbound-proxy ipv4 169.54.5.10 port 1120

172

# Configure domain name abc.com and port number 1 100 of the outbound proxy server for the SIP UA.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] outbound-proxy dns abc.com port 1100

privacy
Description
Use privacy to add the P-Preferred-Identity or P-Asserted-Identity header field. Use undo privacy to remove the configuration. By default, neither the P-Preferred-Identity header field nor the P-Asserted-Identity header field is added.

Syntax
privacy { asserted | preferred } undo privacy

View
SIP client view

Default level
2: System level

Parameters
asserted: Adds the P-Asserted-Identity header field. When the P-Asserted-Identity header field is added, the Privacy header field will be added. The Privacy header field contains the caller identity presentation and screening information, while the P-Asserted-Identity header field contains the caller identity. preferred: Adds the P-Preferred-Identity header field. When the P-Preferred-Identity header field is added, the Privacy header field will be added. The Privacy header field contains the caller identity presentation and screening information, while the P-Preferred-Identity header field contains the caller identity.

Examples
# Add the P-Asserted-Identity header field.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] privacy asserted

proxy
Description
Use proxy to configure proxy server information for a SIP UA. Use undo proxy to remove the proxy server information for a SIP UA. By default, no proxy server information is configured for SIP UA.

Syntax
proxy { dns domain-name | ipv4 ip-address } [ port port-number ]
173

undo proxy { dns | ipv4 }

View
SIP client view

Default level
2: System level

Parameters
dns domain-name: Domain name of the proxy server, which consists of character strings separated by a dot (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.), with a maximum length of 255 characters. ipv4 ip-address: IPv4 address of the proxy server. port port-number: Port number of the proxy server, in the range of 1 to 65535.

Examples
# Configure the IP address 169.54.5.10 and port number 1 120 for the proxy server.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] proxy ipv4 169.54.5.10 port 1120

# Specify the domain name abc.com and port number 1 100 for the proxy server.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] proxy dns abc.com port 1100

reason-mapping pstn
Description
Use reason-mapping pstn to configure PSTN release cause code to SIP status code mappings. Use undo reason-mapping pstn to restore the default. By default, the PSTN release cause code to SIP status code mappings are listed in Table 44. Table 44 Default PSTN release cause code to SIP status code mappings PSTN release cause code
1 2 3 16 17 18

PSTN release cause description

Unallocated (unassigned) number! No route to specified transit network! No route to destination! Normal clearing! User busy! No user responding! 174

SIP status code

404 404 404 --486 408

SIP status description

Not Found Not Found Not Found BYE or CANCEL Busy here Request Timeout

PSTN release cause code

19 20 21 22 23 25 26 27 28 29 31 34 38 41 42 47 55 57 58 63 65

PSTN release cause description

No answer from user! Subscriber absent! Call rejected! Number changed! Redirection to new destination! Exchange routing error! Non-selected user clearing! Destination out of order! Invalid number format (address incomplete)! Facility rejected! Normal, unspecified! No circuit/channel available! Network out of order! Temporary failure! Switching equipment congestion! Resource unavailable, unspecified! Incoming class barred within Closed User Group (CUG)! Bearer capability not authorized! Bearer capability not presently available! Service or option not available, unspecified! Bearer capability not implemented! Only restricted digital information bearer capability is available! Service or option not implemented, unspecified! User not member of Closed User Group (CUG)! Incompatible destination! Recovery on timer expiry! 175

SIP status code

480 480 403 410 410 500 404 502 484 501 480 503 503 503 503 503 403 403 503 500 488

SIP status description

Temporarily unavailable Temporarily unavailable Forbidden Gone Gone Server internal error Not Found Bad Gateway Address incomplete Not implemented Temporarily unavailable Service unavailable Service unavailable Service unavailable Service unavailable Service unavailable Forbidden Forbidden Service unavailable Server internal error Not Acceptable Here

70

488

Not Acceptable Here

79 87 88 102

501 403 503 504

Not implemented Forbidden Service unavailable Gateway timeout

PSTN release cause code

111 127

PSTN release cause description

Protocol error, unspecified! Interworking, unspecified!

SIP status code

500 500

SIP status description

Server internal error Server internal error

Syntax
reason-mapping pstn pstn-code sip sip-code undo reason-mapping pstn pstn-code

View
SIP client view

Default level
2: System level

Parameters
pstn-code: PSTN release cause code, in the range of 1 to 127, but limited to those in Table 44. Because the PSTN release cause code 16 corresponds to a SIP request message, instead of a SIP status code, you can configure no SIP status code for 16. sip-code: SIP status code, in the range of 400 to 699.

Examples
# Map the PSTN release cause code 17 to the SIP status code 408.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice -sip] reason-mapping pstn 17 sip 408

reason-mapping sip
Description
Use reason-mapping sip to configure SIP status code to PSTN release cause code mappings. Use undo reason-mapping sip to restore the default. By default, the SIP status code to PSTN release cause code mappings are listed in Table 45. Table 45 Default SIP status code to PSTN release cause code mappings SIP status code
400 401 402 403 404

SIP status description

Bad Request Unauthorized Payment required Forbidden Not found

PSTN release cause code

41 21 21 21 1

PSTN release cause description

Temporary failure! Call rejected! Call rejected! Call rejected! Unallocated (unassigned) number!

176

SIP status code

405 406 407 408 410 413 414 415 416 420 421 423 480 481 482 483 484 485 486 487 488 500 501 502 503 504 505 513 600

SIP status description

Method not allowed Not acceptable Proxy authentication required Request timeout Gone Request Entity too long Request-URI too long Unsupported media type Unsupported URI Scheme Bad extension Extension Required Interval Too Brief Temporarily unavailable Call/Transaction Does not Exist Loop Detected Too many hops Address incomplete Ambiguous Busy here Request Terminated Not Acceptable here Server internal error Not implemented Bad gateway Service unavailable Server time-out Version Not Supported Message Too Large Busy everywhere

PSTN release cause code

63 79 21 102 22 127 127 79 127 127 127 127 18 41 25 25 28 1 17 127 127 41 79 38 41 102 127 127 17

PSTN release cause description

Service or option not available, unspecified! Service or option not implemented, unspecified! Call rejected! Recovery on timer expiry! Number changed! Interworking, unspecified! Interworking, unspecified! Service or option not implemented, unspecified! Interworking, unspecified! Interworking, unspecified! Interworking, unspecified! Interworking, unspecified! No user responding! Temporary failure! Exchange routing error! Exchange routing error! Invalid number format (address incomplete)! Unallocated (unassigned) number! User busy! Interworking, unspecified! Interworking, unspecified! Temporary failure! Service or option not implemented, unspecified! Network out of order! Temporary failure! Recovery on timer expiry! Interworking, unspecified! Interworking, unspecified! User busy!

177

SIP status code

603 604 606

SIP status description

Decline Does not exist anywhere Not acceptable

PSTN release cause code

21 1 58

PSTN release cause description

Call rejected! Unallocated (unassigned) number! Bearer capability not presently available!

Syntax
reason-mapping sip sip-code pstn pstn-code undo reason-mapping sip sip-code

View
SIP client view

Default level
2: System level

Parameters
sip-code: SIP status code, in the range of 400 to 699, but limited to those in Table 45. pstn-code: PSTN release cause code, in the range of 1 to 127, but limited to those in Table 44.

Examples
# Map the SIP status code to the PSTN release cause code 18.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] reason-mapping sip 486 pstn 18

register-enable
Description
Use register-enable on to enable the SIP registrar. Use register-enable off or undo register-enable to disable the SIP registrar. By default, the SIP registrar is disabled.

Syntax
register-enable { off | on } undo register-enable

View
SIP client view

Default level
2: System level

Parameters
on: Enables the SIP registrar.
178

off: Disables the SIP registrar.

Examples
# Enable the SIP registrar.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] register-enable on

redundancy mode
Description
Use redundancy mode to set the backup mode. Use undo redundancy mode to restore the default backup mode. The default backup mode is parking. Related commands: keepalive.

Syntax
redundancy mode { homing | parking } undo redundancy mode

View
SIP client view

Default level
2: System level

Parameters
homing: Sets the backup mode to homing. parking: Sets the backup mode to parking.

Examples
# Set the backup mode to homing.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] redundancy mode homing

registrar
Description
Use registrar to configure registrar information for the SIP UA. Use undo registrar to remove the registrar information for the SIP UA. By default, no registrar information is configured on the SIP UA. If you execute this command without providing the transport layer protocol type, the UDP protocol will be used during registration; if you execute this command without providing the URL scheme, the SIP URL scheme will be used.

179

The transport layer protocol specified in registrar must have been specified with the listen transport command; otherwise, no register request can be initiated. If TLS is specified in the registrar command, you also need to configure the SSL policy name of the client with the crypto command; otherwise, no register request can be initiated. Before specifying TLS as the transport layer protocol to be used during UA registration with the registrar command, you need to configure the SSL policy name of the client with the crypto command; otherwise, you cannot initiate the register request. You can use this command only when the SIP registration function is disabled.

Syntax
registrar { dns domain-name | ipv4 ip-address } [ port port-number ] [ expires seconds ] [ tcp | tls ] [ scheme { sip | sips } ] [ slave ] undo registrar ipv4 { dns | ipv4 } [ slave ]

View
SIP client view

Default level
2: System level

Parameters
dns domain-name: Domain name of the registrar server, which consists of character strings separated by a dot (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.), with a maximum length of 255 characters. ipv4 ip-address: IPv4 address of the registrar. port port-number: Specifies a port number for the registrar, in the range of 1 to 65535. expires seconds: Specifies the aging time for registration in seconds, in the range of 60 to 65,535. If this value is not provided, the system applies the global registration expiration time set with timer registration expires in SIP client view. tcp: Specifies TCP as the transport layer protocol to be used during UA registration. By default, UDP is adopted. tls: Specifies TLS as the transport layer protocol to be used during UA registration. scheme: Specifies the URL scheme to be used during UA registration. sip: Specifies the SIP scheme as the URL scheme. By default, the SIP URL scheme is adopted. sips: Specifies the SIPS scheme as the URL scheme. slave: Specifies the registrar as a backup server.

Examples
# Configure the IP address 169.54.5.10, the port number 1 120, the registration aging time 120 seconds, and the TCP transport layer protocol for the main registrar.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] registrar ipv4 169.54.5.10 port 1120 expires 120 tcp

180

# Specify the domain name cc.news.com, the port number 1 100, and the registration aging time 120 seconds of the main registrar.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] registrar dns cc.news.com port 1100 expires 120

remote-party-id
Description
Use remote-party-id to add the Remote-Party-ID header field. Use remote-party-id to remove the configuration. By default, the Remote-Party-ID header field is not added.

Syntax
remote-party-id undo remote-party-id

View
SIP client view

Default level
2: System level

Parameters
None

Examples
# Add the Remote-Party-ID header field.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] remote-party-id

reset voice sip connection

Description
Use reset voice sip connection to clear a specified SIP connection over a specific transport layer protocol.

Syntax
reset voice sip connection { tcp | tls } id conn-id

View
User view

Default level
1: Monitor level

Parameters
tcp: Clears a SIP TCP connection.
181

tls: Clears a SIP TLS connection. conn-id: Connection ID, in the range 0 to 1499. You can view connection IDs with the display voice sip connection command.

Examples
# Clear the SIP connection 1 over TCP.
<Sysname> reset voice sip connection tcp id 1

reset voice sip dns-record

Description
Use reset voice sip dns-record to clear SIP DNS records.

Syntax
reset voice sip dns-record

View
User view

Default level
2: System level

Parameters
None

Examples
# Clear SIP DNS lookup records.
<Sysname> reset voice sip dns-record

reset voice sip statistics

Description
Use reset voice sip statistics to clear all the statistics of the SIP client.

Syntax
reset voice sip statistics

View
User view

Default level
2: System level

Parameters
None

Examples
# Clear all the statistics of the SIP client.
<Sysname> reset voice sip statistics

182

rule
Description
Use rule to create an ENUM translation rule. Use undo rule to delete one or all ENUM translation rules. No ENUM translation rule is created by default.

Syntax
rule tag preference value match-pattern replacement-rule domain-name undo rule { tag | all }

View
ENUM translation rule group view

Default level
2: System level

Parameters
tag: Sets the number of the ENUM translation rule in the range 1 to 2147483647. You can configure up to eight ENUM translation rules for the group. preference value: Sets the preference value of the ENUM translation rule in the range 1 to 2147483647. The smaller the value, the higher the priority. match-pattern: Telephone number pattern, supporting regular expressions. It is a string of 1 to 31 characters, which can be numbers and special characters allowed in a regular expression, such as (, ), -, ^, ], {, }, |, *, and +. The - and ^ characters can only be enclosed within brackets [] or braces {}. replacement-rule: Replacement rule, supporting regular expressions. It is a string of 1 to 31 characters, which can be numbers and special character \. domain-name: Domain name. A domain name is a string separated by dots (for example, cc.news.com). A domain name can contain up to 255 case-insensitive characters (including dots), which can be numbers, letters, and special characters - and _. all: Deletes all ENUM translation rules.

Examples
# Create ENUM translation rule 1: the preference is 500, the input telephone number is 01082775326, the translated number is 8277, and the domain name is Beijing.gov. At last, the translated domain name is 7.7.2.8.beijing.gov.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] enum-group 1 [Sysname-voice-dial-enum1] rule 1 preference 500 010(.{4}).* \1 beijing.gov

sip
Description
Use sip to enter SIP client view.
183

Syntax
sip

View
Voice view

Default level
2: System level

Parameters
None

Examples
# Enter SIP client view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip]

sip-comp
Description
Use sip-comp to configure SIP compatibility. Use undo sip-comp to restore the default. By default, The destination number is obtained from the request-line, which is the start line in an SIP request message. The From header field contains the source address and the To header field contains the destination address. The compatibility options are not carried in re-INVITE requests. The Contact header fields of REGISTER messages do not contain the dt parameter.

Syntax
sip-comp { callee | dt | from | t38 | x-parameter } * undo sip-comp { callee | dt | from | t38 | x-parameter } *

View
SIP client view

Default level
2: System level

Parameters
callee: Configures the device to use the destination number in the To header field for sending a SIP request. dt: Configures the Contact header fields of the REGISTER messages to contain the dt parameter. This keyword is used when the device communicates with a VCX device. from: Configures the device to use the address (IP address or DNS domain name) in the To header field as the address in the From header field when sending a SIP request for interoperability with other vendors. By
184

default, the From header field contains the source address and the To header field contains the destination address. t38: When a SIP standard T.38 fax operation is performed, fax parameters T38FaxTranscodingJBIG, T38FaxTranscodingMMR, and T38FaxFillBitRemoval, which are in the SDP fields of the re-INVITE requests and 200 OK responses, do not contain :0. x-parameter: For a fax pass-through operation, the SDP fields of the re-INVITE requests and 200 OK responses contain X-fax description; for a modem pass-through operation, the SDP fields of the re-INVITE requests and 200 OK responses contain X-modem description.

Examples
# Configure the device to use the address in the To field as the address in the From field when sending a SIP request.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] sip-comp from

# Configure the device to use the corresponding event description in the SDP field when sending a re-INVITE request in a fax pass-through or modem-pass-through operation.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] sip-comp x-parameter

sip-comp agent
Description
Use sip-comp agent to configure the User-Agent header field in SIP request messages. Use undo sip-comp agent to remove the configuration. By default, the User-Agent header field in SIP request messages is not configured.

Syntax
sip-comp agent product-name product-version undo sip-comp agent

View
SIP client view

Default level
2: System level

Parameters
agent product-name product-version: Indicates the content of the User-Agent header field in SIP request messages. The product-name and product-version arguments respectively represent the product name and product version of the UAC, each of which is a case-sensitive string of 1 to 31 characters, without { and }.

Examples
Set the User-Agent header field in SIP request messages to company 1.0.
<Sysname> system-view

185

[Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] sip-comp agent company 1.0

sip-comp server
Description
Use sip-comp server to configure the Server header field in SIP response messages. Use undo sip-comp server to remove the configuration. By default, the Server header field in SIP response messages is not configured.

Syntax
sip-comp server product-name product-version undo sip-comp server

View
SIP client view

Default level
2: System level

Parameters
server product-name product-version: Indicates the content of the Server header field in SIP response messages. The product-name and product-version arguments respectively represent the product name and product version of the UAS, each of which is a case-sensitive string of 1 to 31 characters, without { and }.

Examples
Set the Server header field in SIP response messages to company 1.1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] sip-comp server company 1.1

sip-domain
Description
Use sip-domain to configure a domain name for the SIP device. Use undo sip-domain to remove the domain name of the SIP device. By default, IP address, instead of domain name is used.

Syntax
sip-domain domain-name undo sip-domain

View
SIP client view

186

Default level
2: System level

Parameters
domain-name: Domain name of the SIP device. The value consists of 1 to 31 characters, which are not case-sensitive and include numbers 0 through 9, letters A through Z or a through z, underlines (_), hyphens (-), and dots (.).

Examples
# Set the domain name of the SIP device to hello.com.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] sip-domain hello.com

source-bind
Description
Use source-bind to bind the source IP address of SIP packets to an IPv4 address or an interface. Use undo source-bind to remove the binding. By default, the source IP address of SIP packets is not bound, that is, the voice gateway automatically gets an IP address to send out SIP packets.

Syntax
source-bind { media | signal } { interface-type interface-number | ipv4 ip-address } undo source-bind { media | signal }

View
SIP client view

Default level
2: System level

Parameters
media: Media flow. signal: Signaling stream. interface-type interface-number: Specifies an interface. Only Layer 3 Ethernet interfaces, GE interfaces, and dialer interfaces are supported. ipv4 ip-address: IPv4 address to be bound.

Examples
# Bind the IP address 1.1.1.1 to the source IP address of SIP signaling streams.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] source-bind signal ipv4 1.1.1.1

187

timer connection age

Description
Use timer connection age to set the aging time for TCP or TLS connections. Use undo timer connection age to restore the default. By default, the aging time for TCP connections is 5 minutes, and that for TLS connections is 30 minutes.

Syntax
timer connection age { tcp tcp-age-time | tls tls-age-time } undo timer connection age { tcp | tls }

View
SIP client view

Default level
2: System level

Parameters
tcp tcp-age-time: Sets the aging time (in minutes) for TCP connections, in the range 5 to 30. If the idle time of an established TCP connection reaches the specified aging time, the connection will be closed. tls tls-age-time: Sets the aging time (in minutes) for TLS connections, in the range 30 to 180. If the idle time of an established TLS connection reaches the specified aging time, the connection will be closed.

Examples
# Set the aging time for TCP connections to 6 minutes, and that for TLS connections to 60 minutes.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] timer connection age tcp 6 [Sysname-voice-sip] timer connection age tls 60

timer registration retry

Description
Use timer registration retry to set the interval for the voice entity or SIP trunk account to re-register with the registrar after a registration failure. Use undo timer registration retry to restore the default. By default, the interval for the voice entity or SIP trunk account to re-register with the registrar after a registration failure is 240 seconds.

Syntax
timer registration retry seconds undo timer registration retry

View
SIP client view

188

Default level
2: System level

Parameters
seconds: Interval (in seconds) for a voice entity or SIP trunk account to re-register with the registrar after a registration failure, in the range 10 to 3600.

Examples
# Set the interval for the voice entity or SIP trunk account to re-register with the registrar after a registration failure to 300 seconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] timer registration retry 300

timer registration expires

Description
Use timer registration expires to set the registration expiration time. Use undo timer registration expires to restore the default. By default, the registration expiration time is 3600 seconds. Related commands: registrar, registrar server-group, timer registration divider, and timer registration threshold.

Syntax
timer registration expires seconds undo timer registration expires

View
SIP client view

Default level
2: System level

Parameters
Seconds: Registration expiration time in seconds, in the range 60 to 3600.

Examples
# Set the registration expiration time to 600 seconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] timer registration expires 600

timer registration divider

Description
Use timer registration divider to set the registration percentage.
189

Use undo timer registration divider to restore the default. By default, the registration percentage is 80%. Related commands: timer registration expires and timer registration threshold.

Syntax
timer registration divider percentage undo timer registration divider

View
SIP client view

Default level
2: System level

Parameters
percentage: Registration percentage, in the range 50% to 100%.

Examples
# Set the registration percentage to 50%.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] timer registration divider 50

timer registration threshold

Description
Use timer registration threshold to set the lead time before registration. Use undo timer registration-threshold to restore the default. By default, the lead time before registration is 0 seconds. Related commands: timer registration divider and timer registration expires.

Syntax
timer registration threshold seconds undo timer registration threshold

View
SIP client view

Default level
2: System level

Parameters
seconds: Lead time (in seconds) before registration, in the range 0 to 3600.

Examples
# Set the lead time before registration to 100 seconds.
<Sysname> system-view [Sysname] voice-setup

190

[Sysname-voice] sip [Sysname-voice-sip] timer registration threshold 100

timer session-expires
Description
Use timer session-expires to enable periodic refresh of SIP sessions and set the maximum and minimum session expiration time. Use undo timer session-expires to restore the default. By default, the periodic refresh of SIP sessions is not enabled automatically. If periodic refresh of SIP sessions is disabled on the called party but enabled on the calling party, the called party will enable periodic refresh of SIP sessions after negotiation. By default, the minimum session duration is 90 seconds.

Syntax
timer session-expires seconds [ minimum min-seconds ] undo timer session-expires

View
SIP client view

Default level
2: System level

Parameters
seconds: Maximum session expiration time, in the range 90 to 65,535, in seconds. minimum min-seconds: Minimum session expiration time, in the range 90 to 65,535, in seconds.

Examples
# Enable periodic refresh of SIP sessions; set the session expiration time to 1,800 seconds and the minimum session expiration time to 1,000 seconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] timer session-expires 1800 minimum 1000

transport
Description
Use transport to specify the transport layer protocol for outgoing SIP calls. Use undo transport to restore the default. By default, the global transport layer protocol is UDP, and no transport layer protocol is specified for a VoIP voice entity. If the transport layer protocol is not specified for a VoIP voice entity, the global setting is applied. The execution of transport in SIP client view specifies the global transport layer protocol. If you want to configure a different transport layer protocol for an individual call, you can specify the transport layer protocol to be used in corresponding VoIP voice entity view. When the transport layer protocol configured in

191

VoIP voice entity view and that configured in SIP client view are different, the former is adopted. That is, the VoIP voice entity configuration takes precedence over global configuration. This command is effective only when the type of the VoIP voice entity is SIP. The transport layer protocol configured on two communication parties must be the same. That is, if you execute transport tcp on the sender device, you need to execute listen transport tcp on the receiver device. Before specifying TLS as the transport layer protocol, you need to configure the SSL policy names of the client and the server with the crypto command; otherwise, no session request can be initiated.

Syntax
transport { tcp | tls | udp } undo transport

View
SIP client view, VoIP voice entity view

Default level
2: System level

Parameters
udp: Specifies UDP as the transport layer protocol for outgoing SIP calls. tcp: Specifies TCP as the transport layer protocol for outgoing SIP calls. tls: Specifies TLS as the transport layer protocol for outgoing SIP calls.

Examples
# Specify TLS as the transport layer protocol for outgoing SIP calls.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] transport tls

uri
Description
Use uri to configure user information. The user information in the format user-info@domain-name is used to send request messages. Use undo uri to remove the user information. By default, number@SIP-device-domain-name or number @SIP-interface-IP-address is used to send request messages. Related commands: sip-domain.

Syntax
uri user-info [ domain domain-name ] undo uri

View
POTS voice entity view
192

Default level
2: System level

Parameters
user-info: Specifies a user name. A user name contains no more than 31 characters, and can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.). The total length of the user name and the domain name cannot exceed 255 characters. domain domain-name: Specify the domain name. The domain name consists of character strings separated by a dot, for example, aabbcc.com. Each separated string contains no more than 63 characters. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.).The total length of the user name and the domain name cannot exceed 255 characters. If you do not provide the domain name, the domain name configured with sip-domain is used. If sip-domain is not configured, the IP address of the interface is used.

Examples
# Configure user information [email protected] on a POTS voice entity.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] uri user-info hello domain voice.com

url
Description
Use url to specify URL scheme for SIP calls. Use undo user to restore the default. The execution of url in SIP client view specifies the global SIP URL scheme. If you want to configure a different SIP URL scheme for an individual call, you can specify the SIP URL scheme in corresponding VoIP voice entity view. When the SIP URL scheme configured in VoIP voice entity view and that configured in SIP client view are different, the former is adopted. That is, the VoIP voice entity configuration takes precedence over global configuration. By default, SIP URL scheme is adopted. You can use the SIPS scheme only when the transport layer protocol is TLS; otherwise, no session requests will be initiated.

Syntax
url { sip | sips } undo url

View
SIP client view, VoIP voice entity view

Default level
2: System level

Parameters
sip: Specifies SIP as the URL scheme for SIP calls.
193

sips: Specifies SIPS as the URL scheme for SIP calls.

Examples
# Specify SIPS as the global URL scheme for SIP calls.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] url sips

# Configure SIPS as the URL scheme for the SIP calls on VoIP voice entity 1000.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1000 voip [Sysname-voice-dial-entity1000] url sips

user
Description
CAUTION: If realm is configured on the SIP UA, make sure that the value is the same as that configured on the registrar. Otherwise, the SIP UA will fail the authentication due to mismatch. If realm is not configured on a SIP UA, the SIP UA will perform no realm match and consider that the value of realm configured on the registrar is trusted. If it is necessary to configure authentication information in POTS entity view or IVR entity view, the same
authentication information is recommended for the POTS entities or IVR entities configured with the same telephone number.

In the case of authentication, it is forbidden to execute user after the registration function is enabled because this
operation may result in registration update failures.

Use user to configure SIP authentication information. Use undo user to restore the default. By default, the username and password in SIP client view are VOICE-GATEWAY and VOICE-SIP, respectively, while no SIP authentication information is configured in POTS entity view or IVR entity view.

Syntax
user username password { cipher | simple } password [ cnonce cnonce | realm realm ] * undo user

View
SIP client view, POTS entity view, interactive voice response (IVR) entity view

Default level
2: System level

Parameters
username: Username used for registration authentication, a string of 1 to 63 case-sensitive characters. The characters and \ are invalid. cipher: Displays the password of the current user in cipher text mode.
194

simple: Displays the password of the current user in plain text mode. password: Password used for authentication, a case-sensitive string of 1 to 16 characters or 24 characters. When you specify the cipher keyword but enter a password in plain text mode or when you specify the simple keyword, the password may contain 1 to 16 characters. When you specify the cipher keyword and enter a password in cipher text mode, the password must contain 24 characters. cnonce cnonce: Authentication information field used for handshake authentication between the registrar and the SIP UA, This field consists of a string of 1 to 50 case-sensitive characters. The characters and \ are invalid. realm realm: Domain name used for handshake authentication between the registrar and SIP UA. The domain name consists of a string of 1 to 50 case-sensitive characters. The characters and \ are invalid.

Examples
# Configure global SIP authentication information as follows: Username: abcd Password: 1234 Display mode: cipher

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] user abcd password cipher 1234

# Configure SIP authentication information in IVR entity view: Username: abcd Password: 1234 Display mode: cipher

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 100 ivr [Sysname-voice-dial-entity100] user abcd password cipher 1234

wildcard-register enable
Description
Use wildcard-register enable to enable fuzzy telephone number registration. Use undo wildcard-register to disable fuzzy telephone number registration. By default, fuzzy telephone number registration is disabled. When configuring a match template in a POTS entity, you may use a number containing the wildcards of dot (.) and T instead of using a standard E.164 number. After enabling fuzzy telephone number registration, the router retains dots and substitutes asterisks (*) for Ts when sending REGISTER messages. You can use this command only when the SIP registration function is disabled. NOTE: You may use fuzzy telephone number registration only when it is supported on both SIP server and location server.
195

Syntax
wildcard-register enable undo wildcard-register

View
SIP client view

Default level
2: System level

Parameters
None

Examples
# Enable fuzzy telephone number registration.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] wildcard-register enable

196

SIP local survival configuration commands

area-prefix
Description
Use area-prefix to configure an area prefix. Use undo area-prefix to remove an area prefix or all area prefixes. By default, no area prefix is configured. You can configure up to eight area prefixes by repeatedly using the area-prefix command. If multiple area prefixes are configured, the local SIP server adopts the longest match to deal with a called number.

Syntax
area-prefix prefix undo area-prefix { prefix | all }

View
SIP server view

Default level
2: System level

Parameters
prefix: Area prefix, consisting of 1 to 15 digits. all: Removes all area prefixes.

Examples
# Configure two area prefixes.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] area-prefix 8277 [Sysname-voice-server] area-prefix 0108277

authentication
Description
Use authentication to configure authentication information. Use undo authentication to restore the default. By default, no authentication information is configured. If authentication is enabled on the local SIP server, users can successfully register with the local SIP server only after authentication information is configured for them by using the authentication command.

197

Syntax
authentication username username password { cipher | simple } password undo authentication

View
Register user view

Default level
2: System level

Parameters
username username: Username used for authentication, consisting of 1 to 63 case-sensitive characters excluding backslash (\) and double quotation marks (). password password: Password used for registration authentication, consisting of 1 to 16 case-sensitive characters or 24 case-sensitive characters. When you specify the cipher keyword and enter a password in plain text mode or when you specify the simple keyword, the password may contain 1 to 16 characters. When you specify the cipher keyword and enter a password in cipher text mode, the password must contain 24 characters.

Examples
# Configure authentication information.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] register-user 1234 [Sysname-voice-server-user1234] authentication username 1234 password simple 1234

call-route
Description
Use call-route to enter call route view.

Syntax
call-route

View
SIP server view

Default level
2: System level

Parameters
None

Examples
# Enter call route view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] call-route

198

[Sysname-voice-server-route]

call-rule-set
Description
Use call-rule-set to enter call rule set view.

Syntax
call-rule-set

View
SIP server view

Default level
2: System level

Parameters
None

Examples
# Enter call rule set view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] call-rule-set [Sysname-voice-server-set]

srs
Description
Use srs to apply call rule set. Use undo srs to remove the application. By default, no call rule set is applied.

Syntax
srs tag undo srs

View
SIP server view, register user view

Default level
2: System level

Parameters
tag: Call rule set tag, in the range of 0 to 31. The call rule set corresponding to a tag must have been configured.

Examples
# Apply a call rule set in register user view.
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<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] call-rule-set [Sysname-voice-server-set] service 1 [Sysname-voice-server-set-svc0] rule 1 permit outgoing 5... [Sysname-voice-server-set-svc0] rule 2 permit outgoing 1... [Sysname-voice-server-set-svc0] quit [Sysname-voice-server-set] quit [Sysname-voice-server] register-user 1000 [Sysname-voice-server-user1000] srs 1

# Apply a call rule set in sip server view.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] srs 1

display voice sip-server register-user

Description
Use display voice sip-server register-user to display information of registered users, including directory number, registration status, IP address, and port number.

Syntax
display voice sip-server register-user { tag | all } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
tag: Displays the information of the user with the specified tag. all: Displays information of all users. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display information of all users.
<Sysname> display voice sip-server register-user all user number status address ---------------------------------------------------------------------

200

1 2 3

404 325 380

online offline online

192.168.0.98:5060 192.168.0.57:5060

Table 46 Output description Field

User Number Status Address

Description
Tag of a user Directory number of a user Registration status of a user, including

Offline Online
IP address and port number that a user registers

display voice sip-server resource-statistic

Description
Use display voice sip-server resource-statistic to display server resource information.

Syntax
display voice sip-server resource-statistic [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the server resource information.
<Sysname> display voice sip-server resource-statistic SIP Server state: Active CbType SLC_Conf SLC_Call SLC_Sub Total 64 128 64 Used 0 0 0 Free 64 128 64

----------------------------------------------

201

SLC_Reg SSA_Call SSA_Sub

64 128 128

0 0 0

64 128 128

Table 47 Output description Field

SIP Server state CbType Total Used Free SLC_Conf SLC_Call SLC_Sub SLC_Reg SSA_Call SSA_Sub

Description
State of local SIP server:

Active Inactive
Type of the resource control module Total number of resource control modules Number of used resource control modules Number of free resource control modules SLC control module SLC call module SLC subscription module SLC regisration module SSA call module SSA subscription module

expires
Description
Use expires to configure the maximum registration interval. Use undo expires to restore the default. By default, the maximum registration interval is the global active time configured with the server-bind ipv4 command. This command is used to set the maximum registration interval in register user view. When no active time is set for registrations in register user view, the global active time takes effect. When the maximum registration interval configured on the voice gateway is greater than the maximum active time configured on the local SIP server, the maximum registration interval is subject to the one configured on the local SIP server. Related commands: server-bind ipv4.

Syntax
expires time-interval undo expires

View
Register user view

Default level
2: System level
202

Parameters
time-interval: Maximum registration interval in seconds, in the range of 300 to 65535.

Examples
# Set the maximum registration interval for user 1234 to 3,700 seconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] register-user 1234 [Sysname-voice-server-user1234] expires 3700

mode
Description
Use mode to configure the operation mode of the server. Use undo mode to restore the default. By default, the server operates in the alone mode. You can change the operation mode of the server only when the server is disabled. Related commands: server enable.

Syntax
mode { alive-server | alone-server } undo mode

View
SIP server view

Default level
2: System level

Parameters
alive-server: Specifies the server to operate in the alive mode. alone-server: Specifies the server to operate in the alone mode.

Examples
# Specify the server to operate in the alive mode.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysnamevoice-server] mode alive-server

number
Description
Use number to configure the directory number for a registered user. Use undo number to remove the configured directory number.
203

By default, no directory number is configured for the user.

Syntax
number party-number undo number

View
Register user view

Default level
2: System level

Parameters
party-number: Directory number for a registered user, consisting of 1 to 31 digits.

Examples
# Configure the directory number 300 for registered user 1234.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] register-user 1234 [Sysname-voice-server-user1234] number 300

probe remote-server
Description
Use probe remote-server ipv4 to configure the keepalive probe. Use undo probe remote-server ipv4 to remove the keepalive probe. By default, the keepalive probe is not configured. If the local SIP server operates in the alive mode, you can configure probe remote-server ipv4 only when the local SIP server is disabled.

Syntax
probe remote-server ipv4 ipv4-address [ port port-number ] [ keepalive time-interval ] undo probe remote-server ipv4

View
SIP server view

Default level
2: System level

Parameters
ipv4 ipv4-address: IPv4 address of the remote server. port port-number: Port number of the remote server, in the range of 1 to 65535. The default port number is 5060. keepalive time-interval: Interval of sending OPTION messages to the remote server, in seconds, in the range of 64 to 128. The default interval is 64 seconds.
204

Examples
# Configure the keepalive probe.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysnamevoice-server] probe remote-server ipv4 192.168.0.92 keepalive 100

register-user
Description
Use register-user to create a user and enter register user view. Use undo register-user to delete a user or all users. By default, no user is created.

Syntax
register-user tag undo register-user { tag | all }

View
SIP server view

Default level
2: System level

Parameters
tag: Globally unique user tag, in the range of 1 to 2147483647. all: Specifies all user tags.

Examples
# Create user 1234 and enter register user view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] register-user 1234 [Sysname-voice-server-user1234]

rule
Description
Use rule to configure a call rule. Use undo rule to remove a call rule. By default, no call rule is configured.

Syntax
rule tag { deny | permit } { incoming | outgoing } { pattern | any } undo rule { tag | all }
205

View
Service view

Default level
2: System level

Parameters
tag: Call rule tag, in the range of 0 to 31. deny: Denies calls. permit: Permits calls. incoming: Incoming calls. outgoing: Outgoing calls. pattern: Number pattern, consisting of digits and dots (.). Each dot represents a character and can only appear at the end of a number. This argument does not support other characters. any: Any number all: All rules.

Examples
# Configure a call rule.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] call-rule-set [Sysname-voice-server-set] service 1 [Sysname-voice-server-setsvc1] rule 1 deny incoming 2....

service
Description
Use service to create a call rule and enter call rule view. Use undo service to remove a call rule or all call rules. You can use the rule tag { permit | deny } { incoming | outgoing } pattern command in call rule view to set a call rule.

Syntax
service tag undo service { tag | all }

View
Call rule set view

Default level
2: System level

Parameters
tag: Call rule set tag, in the range of 0 to 31.
206

Examples
# Create a call rule.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] call-rule-set [Sysname-voice-server-set] service 1 [Sysname-voice-server-set-svc1]

server-bind ipv4
Description
Use server-bind ipv4 to bind the local SIP server address to the IP address of an interface on the local router. Use undo server-bind ipv4 to remove the binding of the local SIP server address. By default, no IP address is bound, that is, there is no local SIP server. You can configure server- bind ipv4 only when the local SIP server is disabled.

Syntax
server-bind ipv4 ipv4-address [ port port-number ] [ expires time-interval ] undo server-bind ipv4

View
SIP server view

Default level
2: System level

Parameters
ipv4 ipv4-address: IPv4 address. It can be the IP address of an interface on the local router, or the local loopback address 127.0.0.1. Since the local SIP server cannot accept registrations from remote users when the server IP address is set to 127.0.0.1, you are recommended to set the server IP address to the one of an interface on the local router. port port-number: Port number, in the range of 1 to 65535. The default port number is 5060. expires time-interval: Maximum registration interval in seconds, in the range of 300 to 65535. The default interval is 3600 seconds.

Examples
# Bind the interface address 192.168.0.92 to the address of the local SIP server.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] server-bind ipv4 192.168.0.92

server enable
Description
Use server enable to enable the local SIP server.
207

Use undo server enable to disable the local SIP server. By default, the local SIP server is disabled. NOTE: The functions of the local SIP server can take effect only after you configure the server enable command. To configure server enable on the local SIP server operating in the alone mode, you must first configure server-bind
ipv4.

To configure server enable on the local SIP server operating in the alive mode, you must first configure server-bind
ipv4 and probe remote-server ipv4.

Syntax
server enable undo server enable

View
SIP server view

Default level
2: System level

Parameters
None

Examples
# Enable the local SIP server.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] server-bind ipv4 100.1.1.1 [Sysname-voice-server] server enable

sip-server
Description
Use sip-server to enter sip server view.

Syntax
sip-server

View
Voice view

Default level
2: System level

Parameters
None

Examples
# Enter sip server view.
208

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server]

trunk
Description
Use trunk to configure a static route entry. Use undo trunk to delete a static route entry or all static route entries. By default, no call route entry is configured.

Syntax
trunk tag called-number called-pattern ipv4 dest-ip-addr [ port port-number ] [ area-prefix prefix ] undo trunk { tag | all }

View
Call route view

Default level
2: System level

Parameters
tag: Route entry tag, in the range of 0 to 31. Each tag represents a route entry. At most 32 route entries can be configured. called-pattern: Called number pattern, consisting of digits and dots (.). Each dot represents a character and cannot appear before a number. This argument does not support other characters. ipv4 dest-ip-addr: Destination IPv4 address. area-prefix prefix: Area prefix to be added to the route entry which an internal user uses to call an external user, consisting of 1 to 15 digits. all: Deletes all route entries.

Examples
# Configure a static route entry, the destination address is 192.168.0.80, the called number is 1000, and the area prefix is 5000.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] call-route [Sysname-voice-server-route] trunk 20 called-number 1000 ipv4 192.168.0.80 area-prefix 5000

trusted-point
Description
Use trusted-point to specify a trusted node. Use undo trusted-point to delete a trusted node or all trusted nodes.
209

By default, no trusted node is specified. At most eight trusted nodes can be specified on the local SIP server. Only an IP address, rather than a port number, can specify a trusted node.

Syntax
trusted-point ipv4 ipv4-address [ port port-number ] undo trusted-point { ipv4 ipv4-address | all }

View
SIP server view

Default level
2: System level

Parameters
ipv4 ipv4-address: IPv4 address of a trusted node. port port-number: Port number of a trusted node, in the range of 1 to 65535. The default port number is 5060. all: All trusted nodes.

Examples
# Specify a trusted node by its IP address 100.1.1.125.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-server [Sysname-voice-server] trusted-point ipv4 100.1.1.125

210

SIP trunk configuration commands

address
Description
Use address to add a member server to a SIP server group and configure the server information. Use undo address to delete the configuration. By default, a SIP server group has no member server. An index represents the priority of a member server in the SIP server group. The smaller the index value, the higher the priority. You can add at most five member servers to a SIP server group. If an index already exists, the new configuration overwrites the existing one. Related commands: group-name.

Syntax
address index-number { ipv4 ip-address | dns dns-name } [ port port-number ] [ transport { udp | tcp | tls } ] [ url { sip | sips } ] undo address index-number

View
Server group view

Default level
2: System level

Parameters
index-number: Index, in the range 1 to 5. ipv4 ip-address: IPv4 address of the SIP server. dns dns-name: Domain name of the SIP server. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.), with a maximum length of 255 characters. port port-number: Specifies a port number for the SIP server, in the range of 1 to 65535. Without this keyword, the port used depends on the transport layer protocol. In other words, if UDP or TCP is specified as the transport layer protocol, port 5060 is used; if TLS is specified as the transport layer protocol, port 5061 is used. transport: Specifies the transport layer protocol used for the connection between the SIP trunk device and the SIP server. udp: Specifies UDP as the transport layer protocol for the connection between the SIP trunk device and the SIP server. By default, the UDP protocol is adopted. tcp: Specifies TCP as the transport layer protocol for the connection between the SIP trunk device and the SIP server. tls: Specifies TLS as the transport layer protocol for the connection between the SIP trunk device and the SIP server. url: Specifies the URL scheme for the connection between the SIP trunk device and the SIP server.
211

sip: Specifies the SIP scheme as the URL scheme. By default, the SIP URL scheme is adopted. sips: Specifies the SIPS scheme as the URL scheme.

Examples
# Add member server 1 to SIP server group 1, and configure the server information: set the IPv4 address of the SIP server to 192.168.1.1, port number to 20000, and the specify TCP as the transport layer protocol for the connection between the SIP trunk device and the SIP server.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] server-group 1 [Sysname-voice-group1] address 1 ipv4 192.168.1.1 port 20000 transport tcp

address sip server-group

Description
Use address sip server-group to bind a SIP server group to a VoIP voice entity. Use undo address sip server-group to cancel the binding between a SIP server group and a VoIP voice entity. By default, a VoIP voice entity has no any SIP server group bound to it. A VoIP voice entity can have only one existing SIP server group bound to it. Related commands: address sip.

Syntax
address sip server-group group-number undo address sip server-group

View
VoIP voice entity view

Default level
2: System level

Parameters
group-number: Specifies the index of a SIP server group, in the range 1 to 10.

Examples
# Bind SIP server group 1 to VoIP voice entity 1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1 voip [Sysname-voice-dial-entity1] address sip server-group 1

assign
Description
Use assign to assign the host user name or host name allocated by the ITSP to the SIP trunk account. Use undo assign to delete the assigned host user name or host name.
212

By default, no host user name or host name is assigned to a SIP trunk account. You cannot modify or delete the host user name of a SIP trunk account when the account is enabled. You cannot enable the registration function for a SIP trunk account before assigning a host user name for the account. Related commands: register enable.

Syntax
assign { contact-user user-name | host-name host-name } undo assign { contact-user | host-name }

View
Account view

Default level
2: System level

Parameters
contact-user user-name: Host user name, a case-sensitive string of 1 to 31 characters excluding double quotation marks (), backslash (\), or space. host-name host-name: Host name, a string of 1 to 255 characters, which are not case-sensitive. A host name can include letters, digits, hyphens (-), and underscores (_), and cannot include any space.

Examples
# Assign news.com.cn as the host name to SIP trunk account 2.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-trunk account 2 [Sysname-voice-account-2] assign host-name news.com.cn

# Assign 123 as the host user name to SIP trunk account 2.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-trunk account 2 [Sysname-voice-account-2] assign contact-user 123

account enable
Description
Use account enable to enable a SIP trunk account. Use undo account enable to disable a SIP trunk account. By default, a SIP trunk account is enabled. Disabling a SIP trunk account that is already involved in a connection does not delete the connection. In other words, execution of this command takes effect on the next calling of this account.

Syntax
account enable undo account enable
213

View
Account view

Default level
2: System level

Parameters
None

Examples
# Disable SIP trunk account 2.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-trunk account 2 [Sysname-voice-account-2] undo account enable

bind sip-trunk account

Description
Use bind sip-trunk account to bind a SIP trunk account to a VoIP voice entity. Use undo bind sip-trunk account to cancel the binding between a SIP trunk account and a VoIP voice entity. By default, a VoIP voice entity has no any SIP trunk account bound to it. Only an existing SIP trunk account can be bound to a VoIP voice entity. Canceling the binding between a VoIP voice entity and a SIP trunk account that is already involved in a connection does not delete the connection. In other words, execution of this command takes effect on the next calling of this account.

Syntax
bind sip-trunk account account-index undo bind sip-trunk account

View
VoIP voice entity view

Default level
2: System level

Parameters
account-index: Index of the SIP trunk account to be bound to a VoIP voice entity, in the range 1 to 16.

Examples
# Bind SIP trunk account 1 to VoIP voice entity 1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1 voip [Sysname-voice-dial-entity1] bind sip-trunk account 1

214

codec transparent
Description
Use codec transparent to enable codec transparent transmission. Use undo codec transparent to restore the default. By default, codec transparent transmission is disabled, and the SIP trunk device participates in media negotiation between two parties. Enable codec transparent transmission on the VoIP voice entities attached to the internal and external networks.

Syntax
codec transparent undo codec transparent

View
VoIP voice entity view

Default level
2: System level

Parameters
None

Examples
# Enable codec transparent transmission.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1 voip [Sysname-voice-dial-entity1] codec transparent

description
Description
Use description to configure the description for a SIP server group. Use undo description to delete the description for a SIP server group. By default, a SIP server group has no description configured.

Syntax
description text undo description

View
Server group view

Default level
2: System level
215

Parameters
text: Description of a SIP server group, a case-sensitive string of 1 to 80 characters.

Examples
# Configure the description for SIP server group 1 as ITSPA.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] server-group 1 [Sysname-voice-group-1] description ITSPA

display voice sip-trunk account

Description
Use display voice sip-trunk account to display SIP trunk account information.

Syntax
display voice sip-trunk account [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display SIP trunk account information.
<Sysname> display voice sip-trunk account ID 1 2 3 User 1000 2000 3000 Group 1 1 1 Server 202.10.22.188:5060 abc.com:5060 abc.com:5060 Exp 120 400 N/A Status Online Online Logout

Table 48 Output description Field

ID User Group Server

Description
SIP trunk account index Host user name SIP server group index Address or domain name of the registrar

216

Field
Exp

Description
SIP trunk account expiration interval, in seconds If the SIP trunk account is not in the login status, this field is displayed as N/A Registration status of the SIP trunk account:

Status

Disabled Offline Online Login Logout Dnsin: DNS query is being performed before the number is registered Dnsout: DNS query is being performed before the number is deregistered

display voice server-group

Description
Use display voice server-group to display the details of the specified or all SIP server groups.

Syntax
display voice server-group [ group-number ] [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

Parameters
group-number: SIP group server index, in the range 1 to 10. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the details of SIP server group 1.
<Sysname> display voice server-group 1 The information of server group 1 Group name: ITSPA Description: ITSP As Proxy Server list Server list: Index 1: sip:192.169.0.1:5060;transport=udp Index 2: sips:abc.com:5061;transport=tls Current server index: 1 Keepalive mode: Disabled

217

Hot swap mode: Disabled

Table 49 Output description Field

Index

Description
Index of the SIP server group: SIP-URI/SIPS URI; transport layer protocol Keep-alive mode of the SIP server group, including:

Keepalive mode

Disabled REGISTER OPTIONS

Real-time switching function status of the SIP server group, including

Hot swap mode

Disabled Enabled

group-name
Description
Use group-name to specify a name for a SIP server group. Use undo group-name to delete the name of a SIP server group. By default, a SIP server group has no name configured. The name of a SIP server group identifies the SIP server group. The domain name of the carrier server is usually used as the name of a SIP server group. If the name of a SIP server group is not configured, the host name specified in assign is used to identify the group, if any; otherwise, the IP address or domain name of the current server in the SIP server group is used to identify the group. Related commands: address and assign.

Syntax
group-name group-name undo group-name

View
Server group view

Default level
2: System level

Parameters
group-name: Name of a SIP server group, a case-sensitive string of 1 to 31 characters, which can include letters, digits, hyphens (-), underscores (_), and dots (.)

Examples
# Specify ITSP-A as the name for SIP server group 1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] server-group 1

218

[Sysname-voice-group-1] group-name ITSP-A

hot-swap enable
Description
Use hot-swap enable to enable the real-time switching function in a SIP server group. Use undo hot-swap enable to disable the real-time switching function in a SIP server group. By default, the real-time switching function in a SIP server group is disabled.

Syntax
hot-swap enable undo hot-swap enable

View
Server group view

Default level
2: System level

Parameters
None

Examples
# Enable the real-time switching function in SIP server group 1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] server-group 1 [Sysname-voice-group-1] hot-swap enable

keepalive
Description
Use keepalive to enable the keep-alive function and set the keep-alive mode for a SIP server group. Use undo keepalive to disable the keep-alive function for a SIP server group. By default, the keep-alive function for a SIP server group is disabled. With the keep-alive function enabled, the SIP trunk device selects the current server according to the detect result and the redundancy mode. If the keep-alive function is disabled, the current server is always the one with the highest priority in the SIP server group. Related commands: redundancy mode.

Syntax
keepalive { options [ interval seconds ] | register } undo keepalive

View
Server group view

219

Default level
2: System level

Parameters
options: Specifies the OPTIONS keep-alive mode. interval seconds: Interval (in seconds) for sending OPTIONS messages to the SIP servers, in the range 5 to 65535. The default interval is 60 seconds. register: Specifies the REGISTER keep-alive mode.

Examples
# Enable the keep-alive function and set the keep-alive mode for SIP server group 1 to REGISTER mode.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] server-group 1 [Sysname-voice-group-1] keepalive register

match source host-prefix

Description
Use match source host-prefix to match a source host name prefix for a VoIP voice entity. Use undo match source host-prefix to delete the call match rule that specifies the prefix of source host name. By default, no prefix of the source host name is specified as a call match rule for a VoIP voice entity. In other words, all source host names can be matched. The specified prefix of source host name is used to match against the source host names of calls. If the INVITE message received by the SIP trunk device carries the Remote-Party-ID header, the calling information is abstracted from this header field; if the INVITE message received by the SIP trunk device carries the Privacy header, the source host name is abstracted from the P-Asserted-Identity or P-Preferred-Identity header field; if the INVITE message received by the SIP trunk device does not carry any of the above mentioned three header fields, the host name in the From header field of the INVITE message is used as the source host name. You can specify only one source host name prefix based match rule for a VoIP voice entity. If you execute match source host-prefix multiple times, the new configuration overwrites the existing one.

Syntax
match source host-prefix host-prefix undo match source host-prefix

View
VoIP voice entity view

Default level
2: System level

Parameters
Host-prefix: Source host name prefix. The value consists of 1 to 31 characters, which are not case-sensitive and can include letters, digits, underlines (_), hyphens (-), asterisk (*), and dots (.). An asterisk represents a character string of any length, for example, t*m can match the source host names tom, tim, and so on.

220

Examples
# Specify that calls with a source host name starting with Bil are permitted on VoIP voice entity 1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 1 voip [Sysname-voice-dial-entity1] match source host-prefix bil

match destination host-prefix

Description
Use match destination host-prefix to match a destination host name prefix for a VoIP voice entity. Use undo match destination host-prefix to delete the call match rule that specifies the prefix of destination host name. By default, no prefix of destination host name is specified as a call match rule for a VoIP voice entity. In other words, all destination host names can be matched. The specified prefix of destination host name is used to match against the destination host names of calls. The host name in the To header field of an INVITE message received by the SIP trunk device is used as the destination host name. You can specify only one destination host name prefix based match rule for a VoIP voice entity. If you execute match destination host-prefix multiple times, the new configuration overwrites the existing one.

Syntax
match destination host-prefix host-prefix undo match destination host-prefix

View
VoIP voice entity view

Default level
2: System level

Parameters
Host-prefix: Destination host name prefix. The value consists of 1 to 31 characters, which are not case-sensitive and can include letters, digits, underlines (_), hyphens (-), asterisk (*), and dots (.). An asterisk represents a character string of any length, for example, b*y can match the destination host names boy, boundary, and so on.

Examples
# Specify that calls with a destination host name starting with ali are permitted on VoIP voice entity 1.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 3 voip [Sysname-voice-dial-entity3] match destination host-prefix ali

221

match source address

Description
Use match source address to match a source address for a VoIP voice entity. Use undo match source address to delete the call match rule that specifies the source address. By default, no source address is specified as a call match rule for a VoIP voice entity. In other words, all source addresses can be matched. The specified source address is used to match against the source addresses of calls. You can specify only one source address based match rule for a VoIP voice entity. If you execute match source address multiple times, the new configuration overwrites the existing one.

Syntax
match source address { ipv4 ip-address | dns dns-name | server-group group-number } undo match source address

View
VoIP voice entity view

Default level
2: System level

Parameters
ipv4 ip-address: Source IP address. The value must be dotted and can include dots (.), multiplication signs (x), asterisks (*), and digits, where x represents any number between 0 and 9, * represents any number between 0 and 255, and x and * can appear multiple times in one source IP address. Fuzzy matching is supported. For example, 100.1.x.3 indicates any IP address between 100.1.0.3 and 100.1.9.3, and 192.*.*.* indicates any IP address between 192.0.0.1 and 192.255.255.255. dns dns-name: Domain name. A domain name is not case-insensitive and can include letters, digits, hyphens (-), underscores (_), asterisk (*), and dots (.), with a maximum length of 255 characters. If you provide this parameter, the specified domain name is used to match against the source addresses of calls, and a whole-word match is considered a match. For example, if the domain name is configured as sohu, sohu.com is not a match. However, fuzzy matching is supported. An asterisk represents a character string of any length, for example, i*n can match the source addresses ilison, iverson, inn, and so on. server-group group-number: SIP server group index, in the range 1 to 10.

Examples
# Specify that calls with a source IP address in the range 100.1.1.1 to 100.1.1.255 are permitted on VoIP voice entity 3.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 3 voip [Sysname-voice-dial-entity3] match source address ipv4 100.1.1.*

222

proxy server-group
Description
Use proxy server-group to specify a SIP server group to be used as the proxy server. Use undo proxy server-group to delete the proxy server configuration. By default, the system does not use a proxy server to implement SIP message exchange.

Syntax
proxy server-group group-number undo proxy server-group

View
SIP client view

Default level
2: System level

Parameters
group-number: SIP server group index, in the range 1 to 10.

Examples
#Specify SIP server group 5 to be used as the proxy server.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] proxy server-group 5

registrar server-group
Description
Use registrar server-group to associate a SIP trunk account with a SIP server group for registration. Use undo registrar server-group to delete the association between a SIP trunk account and a SIP server group. By default, a SIP trunk account has no SIP server group associated for registration. The specified SIP server group must exist. One SIP trunk account can be associated with only one SIP server group. A SIP trunk account registration cannot be enabled if the account is not associated with any SIP server group. Related commands: register enable and timer registration expires.

Syntax
registrar server-group group-number [ expires seconds ] undo registrar server-group

View
Account view

223

Default level
2: System level

Parameters
group-number: Index of the registrar bound to the SIP trunk account, in the range 1 to 10. expires seconds: Registration expiration interval of a SIP trunk account, in the range 60 to 3600, in seconds. If this parameter is not configured, the system applies the global registration expiration interval configured with timer registration expires in SIP client view.

Examples
# Associate SIP trunk account 1 with SIP server group 2 for registration, and set the registration expiration interval to 300 seconds.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-trunk account 1 [Sysname-voice-account-1] registrar server-group 2 expires 300

register enable
Description
Use register enable to enable the registration function for a SIP trunk account. Use undo register enable to disable the registration function for a SIP trunk account. By default, the registration function for a SIP trunk account is disabled. To enable the registration function for a SIP trunk account, you need to assign it with a host user name or associate it with a SIP server group. When the registration function for a SIP trunk account is enabled, you cannot change its host user name or associated SIP server group. Related commands: assign and registrar server-group.

Syntax
register enable undo register enable

View
Account view

Default level
2: System level

Parameters
None

Examples
# Assign 123 as the host name for SIP trunk account 2, and associate SIP trunk account 2 with SIP server group 2. Then, enable the registration function for the SIP trunk account.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-trunk account 2

224

[Sysname-voice-account-2] assign contact-user 123 [Sysname-voice-account-2] registrar server-group 2 expires 300 [Sysname-voice-account-2] register enable

redundancy mode
Description
Use redundancy mode to configure the redundancy mode for the SIP server group. Use undo redundancy mode to restore the default. By default, the parking redundancy mode is applied. Related commands: keepalive.

Syntax
redundancy mode { homing | parking } undo redundancy mode

View
SIP client view

Default level
2: System level

Parameters
homing: Homing redundancy mode. parking: Parking redundancy mode.

Examples
# Configure the redundancy mode for the SIP server group as homing.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] redundancy mode homing

server-group
Description
Use server-group to create a SIP server group and enter server group view. If the created server group already exits, use this command to enter server group view. Use undo server-group to delete one or all SIP server groups. A SIP server group that is bound to a SIP trunk account or a VoIP voice entity cannot be deleted. The undo server-group all can be executed successfully only when all SIP server groups are not bound to any SIP trunk account or a VoIP voice entity.

Syntax
server-group group-number undo server-group { group-number | all }
225

View
Voice view

Default level
2: System level

Parameters
group-number: SIP server group index, in the range 1 to 10. all: Specifies all SIP server groups.

Examples
# Create SIP server group 1 and enter its view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] server-group 1 [Sysname-voice-group-1]

sip-trunk account
Description
Use sip-trunk account to create a SIP trunk account and enter SIP trunk account view. If the created SIP trunk account already exits, use this command to enter SIP trunk account view. Use undo sip-trunk account to delete one or all SIP trunk accounts. A SIP trunk account that is bound to a SIP server group or a VoIP voice entity cannot be deleted. The undo sip-trunk account all command can be executed successfully only when all SIP trunk accounts are not bound to any SIP server group or a VoIP voice entity. Related commands: bind sip trunk-account.

Syntax
sip-trunk account account-index undo sip-trunk account { account-index | all }

View
Voice view

Default level
2: System level

Parameters
account account-index: SIP trunk account index, in the range 1 to 16. all: Specifies all SIP trunk accounts.

Examples
# Create SIP trunk account 2 and enter its view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-trunk account 2 [Sysname-voice-account-2]

226

sip-trunk enable
Description
Use sip-trunk enable to enable the SIP trunk function. Use undo sip-trunk enable to disable the SIP trunk function. By default, the SIP trunk function is disabled. You are not recommended to use a device enabled with the SIP trunk function as a SIP UA.

Syntax
sip-trunk enable undo sip-trunk enable

View
Voice view

Default level
2: System level

Parameters
None

Examples
# Enable the SIP trunk account.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-trunk enable

user
Description
Use user to configure the authentication user name and password for a SIP trunk account. Use undo user to delete the configured authentication user name and password for a SIP trunk account. By default, a SIP trunk account has no authentication user name or password.

Syntax
user username password { cipher | simple } password undo user

View
Account view

Default level
2: System level

Parameters
username: SIP trunk account username used for registration authentication, a case-sensitive string of 1 to 63 characters. The characters and \ are invalid. simple: Displays the password of the current account in plain text.
227

cipher: Displays the password of the current account in cipher text. password: Password used for authentication, a case-sensitive string of 1 to 16 characters or 24 characters. When you specify the cipher keyword but enter a password in plain text mode or when you specify the simple keyword, the password can contain 1 to 16 characters. When you specify the cipher keyword and enter a password in cipher text mode, the password must contain 24 characters.

Examples
# Configure the authentication user name and password for SIP trunk account 2.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip-trunk account 2 [Sysname-voice-account2] user telA password simple 12345

228

Call services configuration commands

backup-rule loose
Description
Use backup-rule loose to configure the call backup mode as loose. Use undo backup-rule loose to restore the default. By default, the strict call backup mode is applied.

Syntax
backup-rule loose undo backup-rule loose

View
Voice view

Default level
2: System level

Parameters
None

Examples
# Configure the call backup mode as loose.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] backup-rule loose

call-forwarding no-reply enable

Description
Use call-forwarding no-reply enable to enable call forwarding no reply. Use undo call-forwarding no-reply enable to restore the default. By default, call forwarding no reply is disabled. Related commands: call-forwarding on-busy enable, call-forwarding unavailable enable, call-forwarding unconditional enable, and call-forwarding priority. NOTE: This command applies only to FXS voice subscriber lines.

Syntax
call-forwarding no-reply enable forward-number number undo call-forwarding no-reply enable
229

View
Voice subscriber line view

Default level
2: System level

Parameters
forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31 digits, 0 through 9.

Examples
# Enable call forwarding no reply for voice subscriber line 1/0 and set the forwarded-to number to 12345678.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-forwarding no-reply enable forward-number 12345678

call-forwarding on-busy enable

Description
Use call-forwarding on-busy enable to enable call forwarding busy. Use undo call-forwarding on-busy enable to restore the default. By default, call forwarding busy is disabled. Related commands: call-forwarding no-reply enable, call-forwarding unavailable enable, call-forwarding unconditional enable, and call-forwarding priority. NOTE: This command applies only to the FXS voice subscriber line.

Syntax
call-forwarding on-busy enable forward-number number undo call-forwarding on-busy enable

View
Voice subscriber line view

Default level
2: System level

Parameters
forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31 digits, 0 through 9.

Examples
# Enable call forwarding busy for voice subscriber line 1/0 and set the forwarded-to number to 12345678.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-forwarding on-busy enable forward-number 12345678

230

call-forwarding priority
Description
Use call-forwarding priority to configure a priority level for call forwarding. Use undo call-forwarding priority to restore the default. By default, the call forwarding priority level is 2. Related commands: call-forwarding on-busy enable, call-forwarding no-reply enable, call-forwarding unavailable enable, and call-forwarding unconditional enable. NOTE: This command applies only to the FXS voice subscriber line. By default, the priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3 respectively. The
smaller the value is, the higher the priority level is. When you change the priority level of a feature, make sure that different features have different priority levels.

Syntax
call-forwarding priority level undo call-forwarding priority

View
Voice subscriber line view

Default level
2: System level

Parameters
level: Call forwarding priority level, in the range of 1 to 3. The smaller the value, the higher the priority.

Examples
# Configure the call forwarding priority level of 1 for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-forwarding priority 1

call-forwarding unavailable enable

Description
Use call-forwarding unavailable enable to enable call forwarding unavailable. Use undo call-forwarding unavailable enable to restore the default. By default, call forwarding unavailable is disabled. Related commands: call-forwarding on-busy enable, call-forwarding no-reply enable, call-forwarding unconditional enable, and call-forwarding priority. NOTE: This command applies only to the FXS voice subscriber line.

231

Syntax
call-forwarding unavailable enable forward-number number undo call-forwarding unavailable enable

View
Voice subscriber line view

Default level
2: System level

Parameters
forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31 digits, 0 through 9.

Examples
# Enable call forwarding unavailable for voice subscriber line 1/0 and set the forwarded-to number to 12345678.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-forwarding unavailable enable forward-number 12345678

call-forwarding unconditional enable

Description
Use call-forwarding unconditional enable to enable call forwarding unconditional. Use undo call-forwarding unconditional enable to restore the default. By default, call forwarding unconditional is disabled. Related commands: call-forwarding on-busy enable, call-forwarding no-reply enable, call-forwarding unavailable enable, and call-forwarding priority. NOTE: This command applies only to the FXS voice subscriber line.

Syntax
call-forwarding unconditional enable forward-number number undo call-forwarding unconditional enable

View
Voice subscriber line view

Default level
2: System level

Parameters
forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31 digits, 0 through 9.

232

Examples
# Enable call forwarding unconditional for voice subscriber line 1/0 and set the forwarded-to number to 12345678.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-forwarding unconditional enable forward-number 12345678

call-hold enable
Description
Use call-hold enable to enable call hold. Use undo call-hold enable to disable call hold. By default, call hold is disabled. NOTE: This command is only applicable to the FXS voice subscriber line.

Syntax
call-hold enable undo call-hold enable

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable the call hold feature for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-hold enable

call-hold-format
Description
Use call-hold-format to configure a tone playing mode for call hold. Use undo call-hold-format to restore the default. By default, the tone playing mode is inactive, that is, the silent mode.

Syntax
call-hold-format { inactive | sendonly [ media-play media-id ] } undo call-hold-format
233

View
Voice view

Default level
2: System level

Parameters
inactive: Specifies the silent mode for call hold. sendonly: Specifies the playing mode for call hold. media-play media-id: Specifies the ID of the media resource to be displayed, in the range 0 to 2147483647. If you do not specify this keyword, no tones will be played for the called party during call hold.

Examples
# Configure the tone playing mode for call hold as sendonly, and specify the media resource with the ID of 1919 as the played tones.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] media-file g729r8 [Sysname-voice-ivr-g729r8] set-media 1919 cf:/g729/music.wav [Sysname-voice-ivr-g729r8] quit [Sysname-voice-ivr] quit [Sysname-voice] call-hold-format sendonly media-play 1919

call-transfer enable
Description
Use call-transfer enable to enable call transfer. Use undo call- transfer enable to disable call transfer. By default, call transfer is disabled. Related commands: call-transfer start-delay. NOTE: This command applies only to the FXS voice subscriber line. Call hold must be enabled before call transfer.

Syntax
call-transfer enable undo call- transfer enable

View
Voice subscriber line view

Default level
2: System level

234

Parameters
None

Examples
# Enable call transfer for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-transfer enable

call-transfer start-delay
Description
Use call-transfer start-delay to configure a call transfer start delay. Use undo call-transfer start-delay to restore the default call transfer start delay. By default, the call transfer start delay is 3 seconds. Related commands: call-transfer enable. NOTE: This command applies only to the FXS voice subscriber line. Call hold must be enabled before call transfer.

Syntax
call-transfer start-delay number undo call-transfer start-delay

View
Voice subscriber line view

Default level
2: System level

Parameters
number: Call transfer start delay in seconds, in the range of 2 to 5.

Examples
# Set the call transfer start delay to 2 seconds for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-transfer start-delay 2

call-waiting
Description
Use call-waiting to configure parameters for a call waiting tone. Use undo call-waiting to restore the default.

235

By default, two call waiting tones are played once, and if the value of cwi-count number is greater than 1, the interval for playing a call waiting tone is 15 seconds. Related commands: call-waiting enable and call-waiting priority.

Syntax
call-waiting { cwi-count number | cwi-duration length | cwi-interval length } undo call-waiting { cwi-count | cwi-duration | cwi-interval }

View
Voice subscriber line view

Default level
2: System level

Parameters
cwi-count number: Number of a call waiting tone play times, in the range of 1 to 5. cwi-duration length: Number of tones played at one time, in the range of 1 to 3. cwi-interval length: Interval for playing a call waiting tone in seconds, in the range of 10 to 30.

Examples
# Specify a call waiting tone to be played twice for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-waiting cwi-count 2

call-waiting enable
Description
Use call-waiting enable to enable call waiting. Use undo call-waiting enable to disable call waiting. By default, call waiting is disabled. Related commands: call-waiting and call-waiting priority.

Syntax
call-waiting enable undo call-waiting enable

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable call waiting for voice subscriber line 1/0.
236

<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-waiting enable

call-waiting priority
Description
Use call-waiting priority to configure a priority level for call waiting. Use undo call-waiting priority to restore the default. By default, the call waiting priority level is 3. Related commands: call-waiting and call-waiting priority. NOTE: By default, the priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3 respectively. The smaller the value is, the higher the priority level is. When you change the priority level of a feature, make sure that different features have different priority levels.

Syntax
call-waiting priority level undo call-waiting priority

View
Voice subscriber line view

Default level
2: System level

Parameters
level: Call waiting priority level, in the range of 1 to 3. The smaller the value, the higher the priority.

Examples
# Configure the call waiting priority level of 1 for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] call-waiting priority 1

conference enable
Description
Use conference enable to enable the three-party conference function for a voice subscriber line. Use undo conference enable to restore the default. By default, the three-party conference function is disabled.

237

NOTE: The three-party conference function depends on the call hold function. Therefore, you need to enable the call hold
function before configuring three-party conference.

Enabling the three-party conference service in voice subscriber line view will invalidate the local call identification
function. For more information about the configuration of the local call identification function, see distinguish-localtalk in the chapter Voice entity configuration commands.

Syntax
conference enable undo conference enable

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable the three-party conference function for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] conference enable

dialin-restriction enable
Description
Use dialin-restriction enable to enable incoming call barring for a voice subscriber line. Use undo dialin-restriction enable to restore the default. By default, incoming call barring is disabled.

Syntax
dialin-restriction enable undo dialin-restriction enable

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable incoming call barring for voice subscriber line 1/0.
<Sysname> system-view

238

[Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] dialin-restriction enable

dialout-restriction enable
Description
Use dialout-restriction enable to enable outgoing call barring for a voice subscriber line. Use undo dialout-restriction enable to restore the default. By default, outgoing call barring is disabled.

Syntax
dialout-restriction enable password { cipher | simple } password undo dialout-restriction enable

View
Voice subscriber line view

Default level
2: System level

Parameters
password: Sets a password for outgoing call barring. cipher password: Specifies a password in either plain or cipher text mode, and displays it in cipher test mode. A password in plain text must be a string of 1 to 4 decimal digits. A password in cipher text mode must be a string of 24 characters. simple password: Specifies a password in plain text mode and displays it in plain text mode. A password in plain text mode must be a string of 1 to 4 decimal digits.

Examples
# Enable outgoing call barring for voice subscriber line 1/0 and set a password to 1234 in plain text mode.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] dialout-restriction enable password simple 1234

display voice sip subscribe-state

Description
Use display voice sip subscribe-state to display the information of subscription, including phone numbers, subscription server address, effective time, and subscription state. display voice sip subscribe-state [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

239

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the information of subscription.
<Sysname> display voice sip subscribe-state Number 1515 Server Address 100.1.1.101:5060 Expires 3600 Status online +-----------------------------------------------------------------------+

Table 50 Output description Field

Number Server Address Expires

Description
Phone number that proposes the subscription MWI server address, in the format of IP address plus port number or domain name Effective time for the subscription Subscription state:

Status

offline: The subscription has failed online: The subscription has succeeded login: The subscription is being proposed logout: The subscription is being canceled

display voice ss mwi

Description
Use display voice ss mwi to display the information of MWI, including the configuration information of MWI, phone numbers, MWI identifier, number of new messages, number of old messages, number of new urgent messages, number of old urgent messages, total number of general messages, and total number of urgent messages.

Syntax
display voice ss mwi { all | number number } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

Parameters
all: Displays the message waiting indication (MWI) information of all numbers.
240

number number: Displays the MWI information of a specified number. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display all information of MWI.
<Sysname> display voice ss mwi all Message Waiting Indication information: --------------------------------------------------------------------MWI type: Bind MWI server: 100.1.1.101 port:5060 MWI expires: 3600 --------------------------------------------------------------------Number: 1515 Messages-Waiting: Yes Voicemail: 1/3(1/2) Total: 4(3)

Table 51 Output description Field Description

MWI types: MWI type

NoBind-S: Strict match non-binding. NoBind-L: Loose match non-binding. Bind: Binding
MWI server address, in the format of IP address plus port number or domain name. Effective time for the subscription. Phone number.

MWI server MWI expires Number

As shown in the example, number: 1515 indicates that this is the MWI information for number 1515. Message waiting identifier:

Messages-Waiting

Yes: There is/are waiting message(s) on the voice mailbox server. No: There is no waiting message on the voice mailbox server.
As shown in the example, Messages-Waiting: Yes indicates that there are waiting messages in the mailbox of number 1515.

241

Field

Description
Number of new messages/number of old messages (number of new urgent messages/ number of old urgent messages):

Voicemail

As shown in the example, Voicemail: 1/3(1/2) indicates that there are 1 new message, 3 old messages, 1 new urgent message, and 2 old urgent messages in the mailbox, and they can be voice messages, faxes, or mails. Supported message types are determined by the server. Displaying message numbers in alphabet order of their types is not supported. Total number of normal messages (total number of urgent messages).

Total

As shown in the above example, Total: 4(3) indicates that there are 4 normal messages and 3 urgent messages in the mailbox.

feature
Description
Use feature permit to enable the setting of the Feature service. Use undo feature to disable the setting of the Feature service. By default, the setting of the Feature service is disabled. NOTE: This command applies only to the FXS voice subscriber line. The Feature service indicates the service that is used together with the VCX. When you need to interact with the VCX
by using telephone keys, you need to adopt out-of-band NTE transmission to send the DTMF digits to the VCX. The execution of feature permit does not enable out-of-band NTE transmission, and you need to execute outband nte on the called entity to enable it. For more information about the out-of-band NTE transmission, see Voice Configuration Guide.

Syntax
feature { deny | permit } undo feature

View
Voice subscriber line view

Default level
2: System level

Parameters
deny: Disables the setting of the Feature service. permit: Enables the setting of the Feature service.

Examples
# Enable the setting of the Feature service for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] feature permit

242

hunt-group enable
Description
Use hunt-group enable to enable hunt group. Use undo hunt-group enable to disable hunt group. By default, hunt group is disabled. Related commands: hunt-group priority. NOTE: To use the hunt group feature, you need to configure hunt-group enable on all involved FXS voice subscriber lines.

Syntax
hunt-group enable undo hunt-group enable

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable hunt group for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] hunt-group enable

hunt-group priority
Description
Use hunt-group priority to configure a priority level for hunt group. Use undo hunt-group priority to restore the default. By default, the hunt group priority level is 1. Related commands: hunt-group enable. NOTE: By default, the priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3 respectively. The smaller the value is, the higher the priority level is. When you change the priority level of a feature, make sure that different features have different priority levels.

Syntax
hunt-group priority level
243

undo hunt-group priority

View
Voice subscriber line view

Default level
2: System level

Parameters
level: Hunt group priority level, in the range of 1 to 3. The smaller the value, the higher the priority.

Examples
# Configure the hunt group priority level of 2 for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] hunt-group priority 2

joined-conference enable
Description
CAUTION: Enabling the three-party conference service in active participation mode will invalidate the local call identification function (if configured). Use joined-conference enable to enable the three-party conference service in active participation mode for a voice subscriber line. Use undo joined-conference enable to restore the default. By default, the three-party conference in active participation mode is disabled. Related commands: distinguish-localtalk (in the chapter Voice entity configuration commands).

Syntax
joined-conference enable undo joined-conference enable

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable the three-party conference service in active participation mode for voice subscriber line 1/0.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] joined-conference enable

244

mwi enable
Description
Use mwi enable to enable MWI. Use undo mwi enable to disable MWI. By default, MWI is disabled. NOTE: This command is only applicable to the FXS voice subscriber line.

Syntax
mwi enable undo mwi enable

View
Voice subscriber line view

Default level
2: System level

Parameters
None

Examples
# Enable MWI for voice subscriber line 3/0.
<Sysname> system-view [Sysname] subscriber-line 3/0 [Sysname-subscriber-line3/0] mwi enable

mwi tone-duration
Description
Use mwi tone-duration to configure the duration of playing the message waiting tone. Use undo mwi tone-duration to restore the default. By default, the duration of the message waiting tone is 2 seconds. NOTE: This command is only applicable to the FXS voice subscriber line.

Syntax
mwi tone-duration length undo mwi tone-duration

View
Voice subscriber line view

245

Default level
2: System level

Parameters
length: Duration of playing the message waiting tone in seconds, in the range 1 to 60.

Examples
# Configure the duration of the message waiting tone as 4 seconds for voice subscriber line 3/0.
<Sysname> system-view [Sysname] subscriber-line 3/0 [Sysname-subscriber-line3/0] mwi tone-duration 4

mwi-server
Description
Use mwi-server to configure the related information of the voice mailbox server. Use undo mwi-server to remove the configurations. By default, no voice mailbox server information is configured. Before specifying the transport layer protocol with the mwi-server command, you need to configure the same transport layer protocol with the listen transport command; otherwise, no subscription request can be initiated. Before specifying TLS as the transport layer protocol with the mwi-server command, , you need to reference an SSL client policy with the crypto command; otherwise, no subscription request can be initiated.

Syntax
mwi-server { dns domain-name | ipv4 ip-address } [ expires seconds ] [ port port-number ] [ retry seconds ] [ tcp | tls ] [ scheme { sip | sips } ] { bind | no-bind { loose | strict } } undo mwi-server

View
SIP client view

Default level
2: System level

Parameters
dns domain-name: Specifies the domain name of the voice mailbox server, which consists of character strings separated by a dot (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.), with a maximum length of 255 characters. ipv4 ip-address: IP address of the voice mailbox server. expires seconds: Effective time of the subscription in seconds, which is in the range 10 to 72000, and defaults to 3600. port port-number: Port number of the voice mailbox server, which is in the range 1 to 65535. retry seconds: Subscription retry interval in seconds, which is in the range 10 to 7200, and defaults to 120.

246

bind: Binding mode, which indicates that the MWI function is bound with the voice mailbox and the voice mailbox server has set up subscription information for the user agent (UA). Therefore, the UA can receive NOTIFY messages without sending SUBSCRIBEs to the voice mailbox server. no-bind: Non-binding mode, which indicates that the voice mailbox server does not set up subscription information for the UA automatically, so the UA has to send a SUBSCRIBE to the server and after that it can get NOTIFY messages from the server. loose: Loose match, which indicates that strict consistency check is not needed, so the call ID that the NOTIFY is sent to can be different from the call ID that proposed the subscription. strict: Strict match, which indicates that strict consistency check is needed, so the call ID that the NOTIFY is sent to must be the same as the call ID that proposed the subscription. tcp: Specifies TCP as the transport layer protocol to be used during subscription. By default, UDP is adopted. tls: Specifies TLS as the transport layer protocol to be used during subscription. scheme: Specifies the URL scheme to be used during subscription. sip: Specifies SIP as the URL scheme to be used during subscription. sips: Specifies SIPS as the URL scheme to be used during subscription.

Examples
# Configure the IP address of the voice mailbox server as 100.1.1.101, port number as 5060, subscription effective time as 7200 seconds, subscription retry interval as 180 seconds, and the binding mode as bind.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] mwi-server ipv4 100.1.1.101 port 5060 expires 7200 retry 180 bind

# Configure the domain name of the MWI server as cc.hp.com, port number as 5060, subscription effective time as 3600 seconds, subscription retry interval as 240 seconds, and the binding mode as bind.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] mwi-server dns cc.hp.com port 5060 expires 3600 retry 240 bind

# Configure the IP address of the voice mailbox server as 192.168.0.88, transport layer protocol as TCP, and the binding mode as bind.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] mwi-server ipv4 192.168.0.88 tcp bind

timer called-hookon-delay
Description
Use timer called-hookon-delay to enable calling party control and set the on-hook delay time of the called party. Use undo timer called-hookon-delay to restore the default. By default, calling party control is disabled, that is, the on-hook delay of the called party is set to 0.

247

Syntax
timer called-hookon-delay seconds undo timer called-hookon-delay

View
Analog FXS voice subscriber line view

Default level
2: System level

Parameters
seconds: Specifies the on-hook delay of the called party, in seconds, in the range of 0 to 90.

Examples
# Enable calling party control on voice subscriber line 1/0 and set the on-hook delay time of the called party to 90 seconds.
<Sysname> system-view [Sysname] subscriber-line 1/0 [Sysname-subscriber-line1/0] timer called-hookon-delay 90

248

Call-watch configuration commands

The call-watch function is only applicable to voice E1/T1 interfaces. The E1/T1 interfaces mentioned in this document are all voice interfaces.

call-watch group
Description
Use call-watch group to associate the current E1/T1 interface with a call-watch group in the specified mode. Use undo call-watch group to remove the association. By default, an E1/T1 interface is not associated with any call-watch group. NOTE: You can associate an E1/T1 interface with only one call-watch group, and vice versa. You can associate an E1/T1 interface with a call-watch group that has not been created yet but the configuration
does not take effect.

Syntax
call-watch group watch-number { hard | soft } undo call-watch group watch-number [ hard | soft ]

View
E1/T1 interface view

Default level
2: System level

Parameters
watch-number: Specifies a call-watch group number, in the range of 1 to 255. hard: Specifies the call-watch mode as hard. In hard call-watch mode, the E1/T1 interface is set to watch-out state as soon as all the monitored links are detected unavailable regardless of whether calls are present on the interface. An interface in watch-out state does not respond to calls initiated by the connected PBX. soft: Specifies the call-watch mode as soft. In soft call-watch mode, the E1/T1 interface will be set to watch-out state after all the monitored links are detected unavailable if no calls are present on the interface.

Examples
# Associate interface E1 1/0 with monitor group 1 in soft mode.
<Sysname> system-view [Sysname] controller e1 1/0 [Sysname-E1 1/0] call-watch group 1 soft

249

call-watch rule
Description
Use call-watch rule to create a call-watch monitoring rule in a call-watch group. If this rule is the first rule for the call-watch group, the group is created as a result. Use undo call-watch rule to delete the specified monitoring rule, or if no local interface or track object IP is specified, all monitoring rules, from a monitor group. The monitor group is deleted upon removal of the last rule. By default, no call-watch group or call-watch monitoring rule exists. NOTE: A monitor group cannot monitor local interfaces and IP connectivity to remote interfaces at the same time. A monitor group can monitor up to 16 local interfaces or be associated with up to 16 track object IDs associated
with monitored remote IP addresses.

Syntax
call-watch rule watch-number { local interface interface-type interface-number | remote track track-entry-number } undo call-watch rule watch-number [ local interface interface-type interface-number | remote track track-entry-number ]

View
System view

Default level
2: System level

Parameters
watch-number: Specifies a call-watch group number, in the range of 1 to 255. local interface interface-type interface-number: Specifies the type and number of a local interface to be monitored by the call-watch group. remote track track-entry-number: Specifies the track object ID associated with the NQA test group used for monitoring the remote IP address for the track-entry-number argument, in the range 1 to 1024. For more information about NQA and Track configuration commands, see Network Management and Monitoring Command Reference and High Availability Command Reference.

Examples
# Create monitor group 1 and configure it to monitor local interface Ethernet 1/1.
<Sysname> system-view [Sysname] call-watch rule 1 local interface ethernet 1/1

# Create monitor group 2 and associate it with track object ID 1.

<Sysname> system-view [Sysname] call-watch rule 2 remote track 1

250

display call-watch status

Description
Use display call-watch status to display information about the call-watch group associated with the specified E1/T1 interface. If no interface is specified, the information of all call-watch groups associated with an E1/T1 interface is displayed.

Syntax
display call-watch status [ controller controller-type controller-number ] [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
1: Monitor level

Parameters
controller controller-type controller-number: Specifies an E1 or T1 interface by its type and number. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display information of all call-watch groups associated with an E1/T1 interface.
<Sysname> display call-watch status Controller E1 1/0 : UP Call-Watch mode : hard Call-Watch rule 1 remote track 1, state is INVALID Controller E1 1/1 : Watch-Out Call-Watch mode : soft Call-Watch rule 2 local interface Ethernet1/1, network is DOWN

Table 52 Output description Field

Controller E1

Description
The state of the E1 interface associated with a call-watch group, which can be Up, Down, Watch-Out or No-Watch-Out. By setting the interface state to No-Watch-Out, you can disable Call-Watch on the interface. Whether the interface becomes up or down depends on the interface configuration. The operating mode of the call-watch group on the interface, which can be hard or soft. Indicate that monitor group 1 is associated with track object ID 1, which is associated with an NQA test group monitoring a remote IP address.

Call-Watch mode Call-Watch rule 1 remote track 1

251

Field
state is INVALID Call-Watch rule 2 local interface Ethernet1/1 network is DOWN

Description
The state of the track object, which can be POSITIVE, iNVALID, or NEGATIVE. In this example, it is INVALID. Indicate that monitor group 2 monitors local interface Ethernet 1/1. The network layer state of the monitored local interfaces, which can be up or down. It is down in this example.

252

Fax over IP configuration commands

default entity fax
Description
Use default entity fax to set fax parameters to the default values globally. Use undo default entity fax to restore the fax parameters of the system to the defaults. If the call control protocol is SIP, this command can be used only for the originator of the fax request (using private T.38, standard T.38, or fax pass-through protocol). When a fax request is originated using private T.38, standard T.38, or fax pass-through protocol, the fax type is decided according to the configurations. The receiver of the fax request responds to the originator based on the type of the fax request, and then establishes a fax call. NOTE: You must use default entity fax train-mode local to make the configuration made by default entity fax local-train threshold take effect.

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 threshold default entity fax nsf-on default entity fax protocol { standard-t38 | t38 } [ hb-redundancy number | lb-redundancy number ] default entity fax protocol pcm { g71 1alaw | g71 1ulaw } default entity fax train-mode { local | ppp } default entity fax cng-switch enable default entity modem protocol pcm { standard | nte-compatible } { g71 1alaw | g71 1ulaw } undo default entity fax { baudrate | ecm | cng-switch | level | local-train threshold | nsf-on | protocol | train-mode } undo default entity fax cng-switch enable undo default entity modem protocol pcm

View
Voice dial program view

Default level
2: System level

253

Parameters
baudrate: Specifies the maximum fax transmission rate. The inherent default is voice. 2400: Sets the maximum transmission rate to 2400 bps. 4800: Negotiates the baud rate first in accordance with the V.27 fax protocol. The maximum transmission rate is 4800 bps. 9600: Negotiates the baud rate first in accordance with the V.29 fax protocol. The maximum transmission rate to 9600 bps. 14400: Negotiates the baud rate first in accordance with the V.17 fax protocol. The maximum transmission rate to 14,400 bps. disable: Disables fax forwarding.( If the call control protocol is SIP, this keyword disables forwarding of private T.38 and standard T.38 faxes only.) voice: Sets the fax rate to the allowed maximum voice speed for different codec protocols.

cng-switch enable: Enables CNG fax switchover. ecm: Enables the fax error correction mode. It is disabled by default. level level: Specifies the fax signal level in dBm (in the range of 60 to 3). The default value is 15. local-train threshold threshold: Specifies the threshold percentage of fax local training (in the range of 0 to 100). The default value is 10. nsf-on: Enables NSF message transmission. It is disabled by default. protocol: Specifies the transport protocol of the fax. By default, the T.38 fax protocol is applied. Both hb-redundancy number and lb-redundancy number default to 0. standard-t38: Adopts the standard T.38 (UDP) fax protocol, which supports SIP-T.38 protocol. pcm: Enables the pass-through mode. g71 1alaw: Adopts G.71 A-law. 1 g71 1ulaw: Adopts G.71 -law. 1 t38: Enables T.38 fax protocol. hb-redundancy number: Number of redundant high-speed T.38 packets, in the range of 0 to 2. lb-redundancy number: Number of redundant low-speed T.38 packets, in the range of 0 to 5. local: Adopts local training. ppp: Adopts point-to-point training.

train-mode: Specifies the fax training mode. It defaults to ppp.

modem protocol pcm: Specifies a codec type and switching mode for Modem pass-through.

Examples
# Set the maximum fax rate to 9,600 bps globally.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] default entity fax baudrate 9600

254

display voice fax

Description
Use display voice fax statistics to view the statistics of the IP fax module.

Syntax
display voice fax statistics [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the statistics about the FoIP module.
<Sysname> display voice fax statistics Statistics about Fax Session: { Total : 0 0 0 0 0 0 0 : : : : : FAX_VOFR_STANDARD_SWITCH: FAX_VOFR_FRF11_TRUNK FAX_VOFR_FRF11_SWITCH FAX_VOFR_MOTOROLA FAX_VOIP_STDT38 FAX_VOIP_T38 Success : 0 0 0 0 0 0 0 : : : : :

FAX_VOFR_STANDARD_SWITCH: FAX_VOFR_FRF11_TRUNK FAX_VOFR_FRF11_SWITCH FAX_VOFR_MOTOROLA FAX_VOIP_STDT38 FAX_VOIP_T38 Failure : 0

FAX_VOFR_STANDARD_SWITCH: FAX_VOFR_FRF11_TRUNK FAX_VOFR_FRF11_SWITCH : :

0 0 0

255

FAX_VOFR_MOTOROLA FAX_VOIP_STDT38 FAX_VOIP_T38 Last Time : 00:00:00

: : :

0 0 0

FAX_VOFR_STANDARD_SWITCH: FAX_VOFR_FRF11_TRUNK FAX_VOFR_FRF11_SWITCH FAX_VOFR_MOTOROLA FAX_VOIP_STDT38 FAX_VOIP_T38 Processed Pages : 0 : : : : :

00:00:00 00:00:00 00:00:00 00:00:00 00:00:00 00:00:00

FAX_VOFR_STANDARD_SWITCH: FAX_VOFR_FRF11_TRUNK FAX_VOFR_FRF11_SWITCH FAX_VOFR_MOTOROLA FAX_VOIP_STDT38 FAX_VOIP_T38 } : : : : :

0 0 0 0 0 0

Statistics about using fax baudrate: { V27 2400 : V27 4800 : V29 7200 : V29 9600 : V17 7200 : V17 9600 : V17 12000: V17 14400: } Statistics about using ECM or Non-ECM mode: { ECM } Statistics about release reason: { WAIT_DP_BEG_DEMODULATE_TIMEOUT : WAIT_DP_BEG_MODULATE_TIMEOUT WAIT_DP_END_MODULATE_TIMEOUT WAIT_FRAMEACK_TIMEOUT WAIT_T30MSG_PSTN_TIMEOUT WAIT_T30MSG_IP_TIMEOUT : : : : : WAIT_DP_END_DEMODULATE_TIMEOUT : 0 0 0 0 0 0 0 : 0 0 Non-ECM: 0 0 0 0 0 0 0 0

256

SPOOL_TIME_OVER GET_INVALID_T30MESSAGE IPP_CALL_RELEASE NORMAL_RELEASE UNKNOWN_REASON }

: : : : :

0 0 0 0 0

Table 53 Output description Field

FAX_VOFR_STANDARD_SWITCH FAX_VOFR_FRF11_TRUNK FAX_VOFR_FRF11_SWITCH FAX_VOFR_MOTOROLA FAX_VOIP_STDT38 FAX_VOIP_T38 WAIT_DP_BEG_DEMODULATE_TIMEOUT

Description
Fax statistics for standard VoFR Fax statistics for FRF.11 trunk VoFR Fax statistics for FRF.11 switched VoFR Fax statistics for Motorola compatible VoFR Fax statistics for standard T.38 VoIP Fax statistics for T.38 VoIP Statistics of the number of connections released in the case that the DP does not start demodulation within the specified time Statistics of the number of connections released in the case that the DP does not start modulation within the specified time Statistics of the number of connections released in the case that the DP does stop demodulation within the specified time Statistics of the number of connections released in the case that the DP does not stop modulation within the specified time Statistics of the number of connections released in the case that no Frame ACK message is received from the DP within the specified time Statistics of the number of connections released in the case that no T.30 message is received from PSTN within the specified time Statistics of the number of connections released in the case that no T.30 message is received from the IP network within the specified time Statistics of the number of connections released in the case that the number of spooling attempts exceeds the maximum Statistics of the number of connections released owing to invalid T.30 message Statistics of the number of released IPP calls Statistics of the number of connections released normally

WAIT_DP_BEG_MODULATE_TIMEOUT

WAIT_DP_END_DEMODULATE_TIMEOUT

WAIT_DP_END_MODULATE_TIMEOUT

WAIT_FRAMEACK_TIMEOUT

WAIT_T30MSG_PSTN_TIMEOUT

WAIT_T30MSG_IP_TIMEOUT

SPOOL_TIME_OVER

GET_INVALID_T30MESSAGE IPP_CALL_RELEASE NORMAL_RELEASE

257

Field
UNKNOWN_REASON

Description
Statistics of the number of connections released for unknown reasons

fax baudrate
Description
Use fax baudrate to configure the maximum fax baud rate. Use undo fax baudrate to restore the default maximum fax baud rate. If the baud rate is set to a value other than disable and voice, the maximum rate is negotiated first in accordance with the corresponding fax protocol.

Syntax
fax baudrate { 2400 | 4800 | 9600 | 14400 | disable | voice } undo fax baudrate

View
POTS entity view, VoIP entity view, VoFR entity view

Default level
2: System level

Parameters
2400: Sets the maximum fax baud rate to 2,400 bps. 4800: Negotiates the fax baud rate first in accordance with the V.27 fax protocol. The maximum fax baud rate is 4,800 bps. 9600: Negotiates the fax baud rate first in accordance with the V.29 fax protocol. The maximum fax baud rate is 9,600 bps. 14400: Negotiates the fax baud rate first in accordance with the V.17 fax protocol. The maximum fax baud rate is 14,400 bps. disable: Disables the fax function. (If the call control protocol is SIP, private T.38 and standard T.38 faxes are disabled.) voice: Finalizes the allowed maximum fax baud rate first in accordance with voice encoding/decoding protocols. If G.71 is adopted, the fax baud rate is 14,400 bps and the fax protocol is V.17. 1 If G.723.1 Annex A is adopted, the fax baud rate is 4,800 bps and the fax protocol is V.27. If G.726 is adopted, the fax baud rate is 14,400 bps and the fax protocol is V.17. If G.729 is adopted, the fax baud rate is 7,200 bps and the fax protocol is V.29.

Examples
# Configure the gateway to negotiate the fax rate in accordance with the V.29 fax protocol.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 4 pots

258

[Sysname-voice-dial-entity4] fax baudrate 9600

fax cng-switch enable

Description
Use fax cng-switch enable to enable the CNG fax switchover function. Use undo fax cng-switch enable to restore the default. By default, the CNG fax switchover function is disabled.

Syntax
fax cng-switch enable undo fax cng-switch enable

View
POTS entity view, VoIP entity view

Default level
2: System level

Parameters
None

Examples
# Enable the CNG fax switchover function.
<sysmane> system-view [sysname] voice-setup [sysname-voice] dial-program [sysname-voice-dial] entity 100 pots [sysname-voice-dial-entity100] fax cng-switch enable

fax ecm
Description
Use fax ecm to configure the ECM mode for the fax. Use undo fax ecm to restore the default. By default, the ECM mode is not used on the gateway. The fax ecm command is used to perform the forced restriction on the gateway. Only when the fax terminals on both sides support the ECM mode and the gateway uses the ECM mode, the ECM mode will be selected. You must enable the ECM mode for the POTS and VoIP entities of the fax sender and receiver in the ECM mode. NOTE: The configuration of fax ecm in voice entity view is invalid for the FRF.11 trunk mode.

Syntax
fax ecm
259

undo fax ecm

View
POTS entity view, VoFR entity view, VoIP entity view

Default level
2: System level

Parameters
None

Examples
# Configure the gateway to adopt the ECM mode by force.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 4 pots [Sysname-voice-dial-entity4] fax ecm

fax level
Description
Use fax level to configure the transmit energy level of a gateway carrier. Use undo fax level to restore the default. By default, the transmit energy level of a gateway carrier is 15 dBm. Usually, the default transmit energy level of a gateway carrier is acceptable. If fax still cannot be sent when other configurations are correct, try to adjust the transmit energy level.

Syntax
fax level level undo fax level

View
POTS entity view, VoIP entity view, VoFR entity view

Default level
2: System level

Parameters
level: Level of the energy transmitted by a gateway carrier, the transmit energy level attenuation value in dBm, in the range of 60 to 3. The greater the level value is, the higher the energy is. The smaller the level value is, the greater the attenuation is.

Examples
# Configure the transmit energy level of the gateway carrier to 20 dBm.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 4 pots [Sysname-voice-dial-entity4] fax level -20

260

fax local-train threshold

Description
Use fax local-train threshold to configure the fax local training threshold. Use undo fax local-train threshold to restore the default. By default, the fax local training threshold is 10. The point-to-point training means that the gateways do not participate in the rate training between two fax terminals. In this mode, rate training is performed between two fax terminals and is transparent to the gateways. For the point-to-point training, the gateway does not participate in rate training and the threshold is invalid. NOTE: When the local training mode is adopted, the local training threshold configured with fax local-train threshold is valid. When the PPP training mode is adopted, the gateway does not participate in the rate training and the local training threshold is invalid.

Syntax
fax local-train threshold threshold undo fax local-train threshold

View
POTS entity view, VoIP entity view, VoFR entity view

Default level
2: System level

Parameters
threshold: Local training threshold in percentage, in the range of 0 to 100.

Examples
# Configure the percentage of the local training threshold to 20.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] fax local-train threshold 20

fax nsf-on
Description
Use fax nsf-on common to configure the signal transmission mode of fax faculty as a nonstandard mode. Use undo fax nsf-on to restore the default transmission mode. By default, the standard signal transmission mode of fax faculty is adopted.

Syntax
fax nsf-on
261

undo fax nsf-on

View
POTS entity view, VoIP entity view, VoFR entity view

Default level
2: System level

Parameters
None

Examples
# Configure a nonstandard faculty for fax signal transmission.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] fax nsf-on

fax protocol
Description
Use fax protocol to configure the type of protocol used for fax communication with other devices. Use undo fax protocol to restore the default type of protocol used for fax communication with other devices. By default, T.38 negotiation mode is used for fax. If the call control protocol is SIP, this command can be used only for the originator of the fax request. When a fax request is originated, the fax type is decided according to the configurations. The receiver of the fax request responds to the originator based on the type of the fax request, and then establishes a fax call. Low-speed data refers to the V.21 command data, while high-speed data refers to the TCF and image data. To communicate with leading fax terminals in the industry, the standard T.38 protocol must be selected. Likewise, to communicate with other fax terminals supporting a T.38 protocol, the T.38 protocol must be adopted. As the leading devices do not support local training mode for fax, the point-to-point training mode must be adopted in order to implement interworking with the leading devices in 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 pass-through mode is subject to such factors as loss of packet, jitter and delay, so the clocks on both communication sides must be kept synchronized. At present, only G.71 A-law and G.71 law are 1 1 supported, and the VAD function should be disabled.

Syntax
fax protocol { t38 | standard-t38 } [ hb-redundancy number | lb-redundancy number ] fax protocol pcm { g71 1alaw | g71 1law } undo fax protocol

262

View
Voice entity view

Default level
2: System level

Parameters
t38: Uses T.38 fax protocol. With this protocol, a fax connection can be set up quickly. standard-t38: Uses the standard T38 protocol, which supports SIP. lb-redundancy number: The number of low-speed redundant packets. The number argument ranges from 0 to 5, and defaults to 0. hb-redundancy number: The number of high-speed redundant packets. The number argument ranges from 0 to 2, and defaults to 0. pcm: Enables the transparent transmission in the pass-through mode. g71 1alaw: Enables G.71 A-law. 1 g71 1ulaw: Enables G.71 -law. 1

Examples
# Set to 2 the number of high-speed redundant packets sent via the T.38 fax recommendation.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 4 pots [Sysname-voice-dial-entity4] fax protocol t38 hb-redundancy 2

fax train-mode
Description
Use fax train-mode to configure the fax training mode. Use undo fax train-mode to restore the default. By default, the point-to-point mode is adopted. NOTE: VoFR entities only support the PPP training mode.

Syntax
fax train-mode { local | ppp } undo fax train-mode

View
POTS entity view, VoIP entity view, VoFR entity view

Default level
2: System level

263

Parameters
local: Adopts the local training mode. ppp: Adopts the ppp training mode.

Examples
# Configure the local training mode for the gateway.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] fax train-mode local

modem compatible-param
Description
Use modem compatible-param to configure the value of NTE payload type for the NTE-compatible switching mode. Use undo modem compatible-param to restore the default. By default, the value of the NTE payload type is 100. This command is valid only for the NTE-compatible switching mode. Related commands: modem protocol pcm.

Syntax
modem compatible-param payload-type undo modem compatible-param

View
POTS entity view, VoIP entity view

Default level
2: System level

Parameters
payload-type: Value of the NTE payload type for the NTE-compatible switching mode, in the range of 98 to 120.

Examples
# Set the NET payload type to 99 for the NTE-compatible switching mode.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] modem compatible-param 99

264

modem protocol
Description
Use modem protocol pcm to configure the codec type and switching mode for SIP Modem pass-through. Use undo modem protocol to restore the default. By default, SIP Modem pass-through is disabled.

Syntax
modem protocol pcm { standard | nte-compatible } { g71 1alaw | g71 1ulaw } undo modem protocol

View
POTS entity view, VoIP entity view

Default level
2: System level

Parameters
standard: Uses Re-Invite switching for Modem pass-through. nte-compatible: Uses NTE-compatible switching for Modem pass-through. g71 1alaw: Uses g71 1alaw codec for Modem pass-through. g71 1ulaw: Uses g71 1ulaw codec for Modem pass-through.

Examples
# Set the switching mode to NTE-compatible and the codec type to g71 1alaw for SIP Modem pass-through.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] modem protocol pcm nte-compatible g711alaw

reset voice fax statistics

Description
Use reset voice fax statistics to clear IP fax statistics.

Syntax
reset voice fax statistics

View
User view

Default level
2: System level

Parameters
None

265

Examples
# Clear IP fax statistics.
<Sysname> reset voice fax statistics

266

IVR configuration commands

call-normal
Description
Use call-normal to configure the normal secondary-call number match mode for the node. Use undo call-normal to remove the configuration. By default, the match mode of normal secondary-call numbers is not configured.

Syntax
call-normal { length number-length | matching | terminator character } undo call-normal

View
Call node view

Default level
2: System level

Parameters
length number-length: Matches the length of the numbers. The value ranges from 1 to 31. matching: Matches the number. As soon as the matching number is found, the node executes the secondary-call immediately. terminator character: Matches the terminator of the numbers. The value can be any of 0 through 9, pound sign (#), or asterisk (*).

Examples
# Configure node 1 to receive a normal secondary-call number by matching the pound sign (#) as the dial terminator.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 call [Sysname-voice-ivr-node1] call-normal terminator #

# Configure node 1 to receive a normal secondary-call number by matching the length of the number.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 call [Sysname-voice-ivr-node1] call-normal length 7

# Configure node 1 to receive a normal secondary-call number by matching the number.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system

267

[Sysname-voice-ivr] node 1 call [Sysname-voice-ivr-node1] call-normal matching

description
Description
Use description to configure the description string for the node. Use undo description to remove the configuration. By default, no node description string is configured.

Syntax
description text undo description

View
Call node view, Jump node view, Service node view

Default level
2: System level

Parameters
text: Node description string of 1 to 80 case-sensitive characters. Spaces are permitted.

Examples
# Configure the description string for the Jump node.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1] description first-node

display voice ivr call-info

Description
Use display voice ivr call-info to display IVR call information.

Syntax
display voice ivr call-info [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow.
268

exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display IVR call information.
<Sysname> display voice ivr call-info Index 1 2 3 4 Called-Number 101 406 606 806 Caller-Number 100 200 300 400 Entity 101 201 301 401 Node-Id 1 3 6 9 Status PLAY MEDIA WAIT INPUT CALL IDLE -------------------------------------------------------------------------

Table 54 Output description Field

Index Called-Number Caller-Number Entity Node-Id

Description
Index of the call information Number of the called party Number of the calling party IVR voice entity number of the called number Node ID Current status:

Status

IDLE: The node is idle. PLAY MEDIA: The node is playing media files. WAIT INPUT: The node is waiting for the input of the
subscriber.

CALL: The node is calling a number.

display voice ivr media-play

Description
Use display voice ivr media-play to display the IVR playing information.

Syntax
display voice ivr media-play [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide.
269

begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the IVR playing information.
<Sysname> display voice ivr media-play Index 1 2 3 4 Codec g729r8 g711alaw g711ulaw g723r53 Media-Id 1001 1002 1003 1004 Play-Times 3 2 2 2 Status play stop stop stop Type PSTN:1/0 IP:100.1.1.1 IP:100.1.1.1 IP:100.1.1.1 --------------------------------------------------------------------------

Table 55 Output description Field

Index

Description
Playing index Codec type, taking the values:

Codec

g729r8 g71 1alaw g71 1ulaw g723r53

Media-Id Play-Times Status

Media resource file ID Play times of a file Current status:

play stop
Current play type:

PSTN: The called party is from PSTN. In the example,

Type PSTN:1/0 indicates that the called party accesses through the voice subscriber line 1/0.

IP: IP address of the peer media.

display voice ivr media-source

Description
Use display voice ivr media-source to display IVR media resource information.

Syntax
display voice ivr media-source [ | { begin | exclude | include } regular-expression ]

View
Any view

270

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display IVR media resource information.
<Sysname> display voice ivr media-source Codec g729r8 Media-Id 1000 source cfa0:/wav/g7 29r8/0.wav Size (Bytes) 69304 Read-Number 1 Cache-Number 1 --------------------------------------------------------------------------

Table 56 Output description Field

Codec Media-Id Source

Description
Codec type of the media resource file Media resource file ID Media source:

The file name is displayed if the media resource is a file.

Size of the media resource, in bytes.

Size (Bytes) Read-Number Cache-Number

The size of the file is displayed if the media resource is a

file. Number of the read control block Number of the cache

entity ivr
Description
Use entity ivr to create an IVR voice entity and enter IVR voice entity view. Use undo entity ivr to remove the specified IVR voice entity. By default, no IVR voice entity is created. For more information about VoFR, VoIP, and POTS voice entities, see Voice Configuration Guide.

Syntax
entity entity-number ivr undo entity { entity-number | all | ivr }

271

View
Voice dial program view

Default level
2: System level

Parameters
entity-number: Number of an IVR voice entity, in the range 1 to 2147483647. all: All types of voice entities, including VoIP, POTS, VoFR, and IVR voice entities. ivr: Indicates that the voice entity type is IVR.

Examples
#Create IVR voice entity 100 and enter voice entity view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 100 ivr

extension
Description
Use extension to configure an extension secondary-call for a node. You can configure at most ten extension secondary-call numbers for a Call node. Use undo extension to remove the configuration. By default, no extension secondary-call number is configured.

Syntax
extension extension-number call corresponding-number undo extension extension-number

View
Call node view

Default level
2: System level

Parameters
extension-number: Number to be input by the subscriber, a string of 1-31 characters, including 0 through 9, pound sign (#), or asterisk (*). corresponding-number: Extension number, a string of 1-31 characters, including 0 through 9, pound sign (#), or asterisk (*).

Examples
# Configure an extension secondary-call for node 1: when the subscriber dials the number 0, node 1 executes the secondary-call to the number 5000.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system

272

[Sysname-voice-ivr] node 1 call [Sysname-voice-ivr-node1] extension 0 call 5000

input-error
Description
Use input-error to configure the processing method for handling subscriber input errors for a node. Use undo input-error to remove the configuration. By default, no input error processing method is configured for a node.

Syntax
input-error { end-call | goto-pre-node | goto-node node-id } [ media-play media-id [ play-times ] | repeat repeat-times ] * undo input-error

View
Call node view, Jump node view

Default level
2: System level

Parameters
end-call: Terminates the call when the maximum number of input errors is reached. goto-pre-node: Return to the previous node when the maximum number of input errors is reached. goto-node node-id: Jumps to a specified node when the maximum number of input errors is reached. media-play media-id: Specifies the ID of the media resource file to be played after an input error and before the node is executed again, in the range 0 to 2147483647. play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1. repeat repeat-times: Specifies the maximum number of input errors. After an input error occurs, the node will be executed again. When the maximum number of input errors is reached, the system processes according to the configured method. The value of the repeat-times argument ranges from 0 to 255 and defaults to 3.

Examples
# Configure the processing method for handling subscriber input errors for Jump node 1: The node should terminate a call after the maximum number of input errors is reached. The media resource ID is 1000. The node plays voice prompts six times. The maximum number of input errors is five.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1] input-error end-call media-play 1000 6 repeat 5

# Configure the processing method for handling subscriber input errors for Jump node 1:

273

The node should return to the previous node after the maximum number of times permitted for inputting errors is reached. The media resource ID is 1001. The node plays voice prompts only once. The maximum number of input errors is three.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1] input-error goto-pre-node media-play 1001 1 repeat 3

# Configure the processing method for handling subscriber input errors for Jump node 1: The node should jump to node 20 after the maximum number of times permitted for inputting errors is reached. The media resource ID is 1002. The node plays voice prompts three times. The maximum number of input errors is five.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr ] node 1 jump [Sysname-voice-ivr-node1 ] input-error goto-node 20 media-play 1002 3 repeat 5

ivr-input-error
Description
Use ivr-input-error to configure processing method for handling subscriber input errors globally. Use undo ivr-input-error to restore the default. By default, the maximum number of input errors is three. The system does not play voice prompts for input errors and terminates the call after the maximum number of input errors is reached.

Syntax
ivr-input-error { media-play media-id [ play-times ] | repeat repeat-times } * undo ivr-input-error

View
IVR management view

Default level
2: System level

Parameters
media-play media-id: Specifies the ID of the media resource file to be played after an input error occurs and before the node is executed again. The value ranges from 0 to 2147483647. play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1.

274

repeat repeat-times: Specifies the maximum number of input errors. After an input error occurs, the node will be executed again. When the maximum number of input errors is reached, the system terminates the call. The value of the repeat-times argument ranges from 0 to 255 and defaults to 3.

Examples
# Configure the global processing method for handling subscriber input errors: The media resource ID is 1002. The node plays voice prompts twice. The maximum number of input errors is five.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] ivr-input-error media-play 10002 2 repeat 5

ivr-root
Description
Use ivr-root to specify the root node (the first node to be executed) of an IVR voice entity. Use undo ivr-root to remove the configuration. By default, the root node is not configured for an IVR voice entity.

Syntax
ivr-root node-id undo ivr-root

View
IVR voice entity view

Default level
2: System level

Parameters
node-id: Specifies the ID of the root node, in the range 1 to 256.

Examples
# Configure the root node of IVR voice entity 100.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 100 ivr [Sysname-voice-dial-entity100] ivr-root 1

ivr-system
Description
Use ivr-system to enter IVR management view.

275

Syntax
ivr-system

View
Voice view

Default level
2: System level

Parameters
Node

Examples
# Enter IVR management view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr]

ivr-timeout
Description
Use ivr-timeout to configure the IVR global input-timeout processing method. Use undo ivr-timeout to restore the default. By default, the timeout time is 10 seconds, and the maximum timeout times are three. The system does not play voice prompts for input timeouts and terminates the call after the maximum number of times is reached.

Syntax
ivr-timeout { expires seconds | media-play media-id [ play-times ] | repeat repeat-times } * undo ivr-timeout

View
IVR management view

Default level
2: System level

Parameters
expires seconds: Specifies the timeout time. The value ranges from 1 to 255 and defaults to 10, in seconds. media-play media-id: Specifies the ID of the media resource file to be played after an input timeout occurs and before the node is executed again. The value ranges from 0 to 2147483647. play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1. repeat repeat-times: Specifies the maximum number of input timeouts. After an input timeout occurs, the node will be executed again. When the maximum number of input timeouts is reached, the system terminates the call. The value of the repeat-times argument ranges from 0 to 255 and defaults to 3.

Examples
# Configure the global input timeout processing method:
276

The timeout time is 20 seconds. The media resource ID is 100001. The node plays voice prompts only once. The maximum number of timeout is twice

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] ivr-timeout expires 20 media-play 100001 1 repeat 2

media-file
Description
Use media-file to enter voice media resource management view. Related commands: ivr-system and set-media.

Syntax
media-file { g71 1alaw | g71 1ulaw | g723r53 | g729r8 }

View
IVR management view

Default level
2: System level

Parameters
g71 1alaw: Enters g71 1alaw codec view. g71 1ulaw: Enters g71 1ulaw codec view. g723r53: Enters g723r53 codec view. g729r8: Enters g729r8 codec view.

Examples
# Enter g729r8 codec view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] media-file g729r8 [Sysname-voice-ivr-g729r8]

media-play
Description
Use media-play to specify the audio file that will be played to the subscriber when the node is waiting for the subscriber to press keys. Use undo media-play to restore the default. By default, the audio file that will be played to the subscriber when the node is waiting for the subscriber to press keys is not configured.
277

Syntax
media-play media-id [ play-times ] [ force ] undo media-play

View
Call node view, Jump node view

Default level
2: System level

Parameters
media-id: Media resource file ID, in the range 0 to 2147483647. play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1. force: Specifies that the subscriber can press the key only after the play of voice prompts is finished, otherwise, subscriber input is considered invalid. By default, the force keyword is not specified, that is,. subscriber input is valid during voice prompt display.

Examples
# Specify the node to play the audio file 10000 three times to the subscriber when waiting for the subscriber to press keys.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1] media-play 10000 3 force

node
Description
Use node to create an IVR voice entity node or enter the view of an existing node. Use undo node to delete a specified or all IVR nodes.

Syntax
node node-id { call | jump | service } undo node { node-id | all }

View
IVR management view

Default level
2: System level

Parameters
node-id: Specifies the node ID, in the range 1 to 256. call: Creates a Call node, which executes a secondary-call after the subscriber inputs a number. jump: Creates a Jump node, which jumps to another node according to the input of the subscriber. service: Creates a Service node, which executes various operations, such as playing audio files, jumping, executing immediate secondary-call, terminating a call, and playing voice prompts.
278

all: All types of nodes.

Examples
# Create Jump node 1 and enter its view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1]

operation
Description
Use operation to configure function for a Service node. Use undo operation to remove the configuration. By default, no function is configured for a Service node. If an executed function is to jump to another node or to terminate a call, the rest one or two functions will not be executed. Related commands: select-rule operation-order.

Syntax
operation number { call-immediate call-number | end-call | goto-node node-id | goto-pre-node | media-play media-id [ play-times ] } undo operation number

View
Service node view

Default level
2: System level

Parameters
number: Specifies the serial number of the configured function, in the range 1 to 3. call-immediate call-number: Indicates immediate secondary-call. The call-number argument represents the phone number of the secondary-call. end-call: Terminates a call. goto-node node-id: Jumps to a specified node. The node-id argument represents the node ID, in the range 1 to 256. goto-pre-node: Returns to the previous node. media-play media-id: Specifies the media resource file ID, in the range 0 to 2147483647. play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1.

Examples
# Configure Service node 1.
<Sysname> system-view [Sysname] voice-setup

279

[Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 service [Sysname-voice-ivr-node1] operation 1 end-call

select-rule operation-order
Description
Use select-rule operation-order to specify the execution order of the configured functions. Use undo select-rule operation-order to restore the default. By default, the execution order is select-rule operation-order 1 2 3. Related commands: operation.

Syntax
select-rule operation-order 1st-operation 2nd-operation 3rd-operation undo select-rule operation-order

View
Service node view

Default level
2: System level

Parameters
1st-operation: Specifies the serial number of the function to be executed first. The value ranges from 1 to 3. 2nd-operation: Specifies the serial number of the function to be executed secondly. The value ranges from 1 to 3, and cannot be the same as the value of 1st-operation. 3rd-operation: Specifies the serial number of the function to be executed thirdly. The value ranges from 1 to 3, and cannot be the same as the value of 1st-operation and 2nd-operation.

Examples
# Specify the execution order of the configured functions for node 1 as 1->3->2.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 service [Sysname-voice-ivr-node1] select-rule operation-order 1 3 2

set-media
Description
Use set-media to specify a media resource ID for a media resource file. Each codec can be configured with up to 256 media resource IDs. Use undo set-media to remove the configuration. By default, no customized media ID is specified for a media resource file. Related commands: media-file.

280

Syntax
set-media media-id file filename undo set-media { media-id | all }

View
Voice media resource management view

Default level
2: System level

Parameters
media-id: Specifies the media resource file ID, in the range 1000 to 2147483647. file filename: Media resource file name. Spaces are permitted, and the file name must be in double-quote marks. The maximum length of the value is 136 bytes, excluding the length of double-quote marks. all: All media resource file IDs.

Examples
# Specify 10001 as the media resource ID of the media resource file cfa0:/g729/ring.wav.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] media-file g729r8 [Sysname-voice-ivr-g729r8] set-media 10001 file cfa0:/g729/ring.wav

timeout
Description
Use timeout to configure the input timeout processing method for an IVR node. Use undo timeout to remove the configuration. By default, no input timeout processing method is configured for an IVR node.

Syntax
timeout { end-call | goto-pre-node | goto-node node-id } [ expires seconds | media-play media-id [ play-times ] | repeat repeat-times ] * undo timeout

View
Call node view, Jump node view

Default level
2: System level

Parameters
end-call: Terminates the call when the maximum number of input timeouts is reached. goto-pre-node: Return to the previous node when the maximum number of input timeouts is reached. goto-node node-id: Jumps to a specified node when the maximum number of input timeouts is reached. The value ranges from 1 to 256.
281

expires seconds: Specifies the timeout time. The value ranges from 1 to 255 and defaults to 10, in seconds. media-play media-id: Specifies the ID of the media resource file to be played after an input timeout occurs and before the node is executed again. The value ranges from 0 to 2147483647. play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1. repeat repeat-times: Specifies the maximum number of input timeouts. After an input timeout occurs, the node will be executed again. When the maximum number of input timeouts is reached, the system terminates the call. The value of the repeat-times argument ranges from 0 to 255 and defaults to 3.

Examples
# Configure the input timeout processing method for Jump node 1: The node should terminate the call after the maximum number of times permitted for input timeouts is reached. The maximum number of input timeouts is three.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1] timeout end-call repeat 3

user-input
Description
Use user-input to configure the node to execute the jump operation based on the input of the subscriber. Use undo user-input to remove the configuration. By default, no jump operation is configured. You can configure up to 12 operations for a Jump node.

Syntax
user-input character { end-call | goto-node node-id | goto-pre-node } undo user-input character

View
Jump node view

Default level
2: System level

Parameters
character: Input of the subscriber. The value can be any of 0 through 9, pound sign (#), or asterisk (*). end-call: Terminates the call. goto-node node-id: Jumps to the specified node. The value node-id ranges from 1 to 256. goto-pre-node: Return to the previous node.

Examples
# Configure the node to terminate the call if the subscriber presses 0.
<Sysname> system-view

282

[Sysname] voice-setup [Sysname-voice] ivr-system [Sysname-voice-ivr] node 1 jump [Sysname-voice-ivr-node1] user-input 0 end-call

283

VoFR configuration commands

address
Description
Use address to configure a channel to the peer voice gateway. Use undo address to remove the configuration. By default, no channel to the peer voice gateway is configured. The FRF.1 sub-channel number to be configured must be available; the FRF.1 sub-channel is not occupied. 1 1 A voice channel will be established for the VoFR entity immediately you execute the address vofr-static command. The voice channel will be removed after you execute the undo form of the command or delete the VoFR entity. Related commands: call-mode, vofr, trunk-id, and display fr vofr-info.

Syntax
address { vofr-dynamic serial interface-number dlci-number | vofr-static serial interface-number dlci-number cid-number } undo address { vofr-dynamic | vofr-static }

View
VoFR entity view

Default level
2: System level

Parameters
vofr-dynamic: Specifies a VoFR entity to adopt the dynamic call mode. vofr-static: Specifies a VoFR entity to adopt the FRF.1 trunk mode. 1 serial interface-number: Specifies the destination interface of a VoFR entity. dlci-number: Destination virtual circuit number of a VoFR entity, in the range of 16 to 1007. cid-number: Destination FRF.1 sub-channel number of a VoFR entity, in the range of 4 to 255. 1

Examples
# Specify DLCI 100 to adopt the dynamic call mode.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 4 vofr [Sysname-voice-dial-entity4] match-template 12345 [Sysname-voice-dial-entity4] address vofr-dynamic serial1/0 100

284

call-mode
Description
Use call-mode to configure the mode in which calls between the VoFR entity and the peer voice entity are established. Use undo call-mode to restore the default call mode. By default, the dynamic mode is adopted. Dynamic call mode: When a call is originated, the frame relay will randomly select an idle FRF.1 1 sub-channel to establish a voice channel. After the call is completed, the frame relay will immediately remove the voice channel and release the corresponding FRF.1 sub-channel. The call control protocol 1 used in the dynamic call mode is specified by executing vofr in interface DLCI view. FRF.1 trunk mode: A voice channel is established when you execute the address vofr-static command. 1 The voice channel is directly used to establish calls. After the call is completed, the voice channel remains until it is manually cleared. In the FRF.1 trunk mode, you must use trunk-id to configure a 1 PSTN-dailed number for the terminating VoFR entity.

Related commands: trunk-id and address.

Syntax
call-mode { dynamic | static } undo call-mode

View
VoFR entity view

Default level
2: System level

Parameters
dynamic: Adopts the dynamic call mode. static: Adopts the FRF.1 trunk mode. 1

Examples
# Configure the FRF.1 trunk mode for VoFR entity 10. 1
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 vofr [Sysname-voice-dial-entity10] call-mode static

cid select-mode
Description
Use cid select-mode to configure the CID selection mode which the originating side of a VoFR call adopts. Use undo cid select-mode to restore the default. By default, CIDs are cyclically selected in descending order.

285

In the dynamic mode, it is possible that multiple voice channels share one DLCI. The same CID at both ends may lead to a call collision. To prevent call collisions, you may configure different CID selection modes at both ends. Related commands: vofr.

Syntax
cid select-mode { max-poll | min-poll } undo cid select-mode

View
Interface DLCI view

Default level
2: System level

Parameters
max-poll: Selects circuit IDs cyclically in descending order. min-poll: Selects circuit IDs cyclically in ascending order.

Examples
# Set the CID selection mode to min-poll on DLCI 100.
<Sysname> system-view [Sysname] interface serial 1/0 [Sysname-Serial1/0] fr dlci 100 [Sysname-fr-dlci-100] cid select-mode min-poll

display fr vofr-info
Description
Use display fr vofr-info to display the FRF.1 sub-channel information on a VoFR DLCI. You can use the 1 display fr vofr-info serial interface-number command to display the FRF.1 sub-channel information on a 1 specified interface sub-interface. The information of all FRF.1 sub-channels will be displayed if no interface 1 sub-interface is specified. You can use the display fr vofr-info dlci-number to display the FRF.1 sub-channel 1 information on a specified DLCI.

Syntax
display fr vofr-info [ serial interface-number [ dlci-number ] ] [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: Monitor level

Parameters
serial interface-number: Displays the FRF.1 sub-channel information on a specified interface. 1 dlci-number: Virtual circuit number, in the range of 16 to 1007 |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide.
286

begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the FRF.1 sub-channel information on a VoFR DLCI. 1
<Sysname> display fr vofr-info interface(dlci) Serial2/0:0(100) Serial2/0:0(100) vofr-mode vofr-nonstandard vofr-nonstandard cid 5 4 cid-type frag-data voice-signal

# Display the FRF.1 sub-channel information on the specified interface. 1

<Sysname> display fr vofr-info ser2/0:0 17 interface(dlci) Serial2/0:0(17) Serial2/0:0(17) vofr-mode vofr-nonstandard vofr-nonstandard cid 254 255 cid-type frag-data voice-signal pkts(in/out/drop) 0/0/0 0/1068/0

Table 57 Output description Field

interface(dlci) vofr-mode cid cid-type pkts(in/out/drop)

Description
Frame relay interface name (DLCI number) VoFR call control protocol, for example, VoFR nonstandard-compatible and VoFR-Huawei-compatible. Voice channel number Type of a voice channel Numbers of inbound, outbound and dropped packets

entity vofr
Description
Use entity vofr to enter VoFR entity view. Use undo entity vofr to remove the existing voice entity. When you configure VoIP entities, POTS entities, VoFR entities, and IVR entities, they should be identified with different entity-number. For more information about IVR, VoIP, and POTS voice entities, see Voice Configuration Guide.

Syntax
entity entity-number vofr undo entity { entity-number | all | vofr }

View
Voice dial program view

Default level
2: System level
287

Parameters
entity-number: Entity number, in the range of 1 to 2147483647. all: All types of voice entities, including VoIP, POTS, VoFR, and IVR voice entities. vofr: VoFR voice entity

Examples
# Create a VoFR entity and number it 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 vofr

outband vofr
Description
Use outband vofr to configure the out-of-band DTMF transmission mode. Use undo outband to restore the default. By default, the inband DTMF transmission mode is adopted.

Syntax
outband vofr undo outband

View
VoFR entity view

Default level
2: System level

Parameters
None

Examples
# Configure the out-of-band DTMF transmission mode for VoFR entity 10.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 vofr [Sysname-voice-dial-entity10] outband vofr

seq-number
Description
Use seq-number to configure the VoFR packets sent by the local voice gateway to carry a sequence number. Use undo seq-number to restore the default. By default, the VoFR packets sent by the local voice gateway do not carry any sequence number.
288

NOTE:
Usually, the configuration of the originating voice gateway determines whether VoFR packets carry a sequence number. Routers of some manufacturers do not comply with the above rule, but force VoFR packets to carry a sequence number when a specific codec is adopted. If a call failure or severe voice distortion occurs when the device is interconnected with a router of a third party, you can try making VoFR packets carry a sequence number. The terminating voice gateway can determine whether any voice packet loss, duplicate voice packet, or out-of-sequence occurs according to sequence numbers, which helps compensate voice. However, the use of sequence numbers will increase the required network bandwidth. Therefore, you can determine whether to use sequence numbers according to the actual condition.

Syntax
seq-number undo seq-number

View
VoFR entity view

Default level
2: System level

Parameters
None

Examples
# Configure voice packets sent by VoFR entity 10 to carry a sequence number.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 vofr [Sysname-voice-dial-entity10] seq-number

timestamp
Description
Use timestamp to configure VoFR packets sent by the local voice gateway to carry a timestamp. Use undo timestamp to restore the default. By default, the VoFR packets sent by the local voice gateway do not carry any timestamp.

Syntax
timestamp undo timestamp

View
VoFR entity view

Default level
2: System level
289

Parameters
None

Examples
# Configure voice packets sent by VoFR entity 10 to carry a timestamp.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 vofr [Sysname-voice-dial-entity10] timestamp

trunk-id
Description
Use trunk-id to configure a PSTN-dialed number in the FRF.1 trunk mode. 1 Use undo trunk-id to restore the default. By default, no PSTN-dialed number is configured in the FRF.1 trunk mode. 1 Related commands: call-mode.

Syntax
trunk-id string undo trunk-id

View
VoFR entity view

Default level
2: System level

Parameters
string: PSTN-dialed number, a string of 1 to 31 characters.

Examples
# Configure the PSTN-dialed number 3333 for VoFR entity 2222 in the FRF.1 trunk mode. 1
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 2222 vofr [Sysname-voice-dial-entity2222] call-mode static [Sysname-voice-dial-entity2222] trunk-id 3333

voice bandwidth
Description
Use voice bandwidth to reserve a VoFR voice bandwidth. Use undo voice bandwidth to remove the reserved bandwidth. By default, no bandwidth is reserved for voice.
290

This command is configured in frame relay class view and takes effect only after the DLCI references such a frame relay class. Otherwise, no voice bandwidth will be available and call setup will fail.

Syntax
voice bandwidth reserved-bps [ reserved ] undo voice bandwidth

View
Frame relay class view

Default level
2: System level

Parameters
reserved-bps: Reserved voice bandwidth in bps, in the range of 8,000 to 45,000,000. reserved: Reserves a VoFR voice bandwidth.

Examples
# Reserve a maximum bandwidth of 8 kbps for voice in frame relay class test1 view
<Sysname> system-view [Sysname] fr class test1 [Sysname-fr-class-test1] voice bandwidth 8000 reserved

vofr
Description
Use vofr to configure a VoFR operation mode for a DLCI. Use undo vofr to restore the default. By default, no VoFR operation mode is configured. If the VoFR operation mode is set to Motorola-compatible and the call mode is set to static (FRF.1 trunk mode), 1 a call failure will occur. In the Huawei-compatible or Motorola-compatible mode, the T1.167 Annex G protocol is adopted. In this case, different ANNEX G-compliant control block types must be configured at both ends: one to DTE and the other to DCE. Related commands: call-mode.

Syntax
vofr { huawei-compatible [ dce | dte ] | motorola-compatible [ dce | dte ] | nonstandard-compatible signal-channel ccid-no data-channel dcid-no [ keepalive ] } undo vofr

View
Interface DLCI view

Default level
2: System level

291

Parameters
huawei-compatible: Adopts the Huawei-compatible mode. motorola-compatible: Adopts the Motorola-compatible mode for compatibility with VoFR of Motorola routers. The FRF.1 trunk mode does not support the Motorola-compatible protocol. 1 dce: Specifies the virtual circuit to serve as a DCE in compliance with Annex G. dte: Specifies the virtual circuit to serve as a DTE in compliance with Annex G. nonstandard-compatible: Adopts the nonstandard-compatible mode for compatibility with VoFR of Cisco routers. signal-channel ccid-no data-channel dcid-no: FRF.1 sub-channel numbers respectively used by signaling 1 and data when VoFR operates in the nonstandard-compatible mode, in the range of 4 to 255. keepalive: Sends KeepAlive messages regularly. In the nonstandard-compatible mode, KeepAlive messages are regularly sent so as to monitor and control the sub-channel status. If the keepalive keyword is configured, network congestion is considered occurring when one end fails to receive any KeepAlive message within a period of time. In this case, the active call control sub-channel will be deactivated, and no voice call can be set up any longer. If the keepalive keyword is not configured, the control sub-channel status is synchronized with the PVC status.

Examples
# Set the call control protocol on DLCI 1000 to nonstandard-compatible, call control sub-channel number (ccid) to 4, and data sub-channel (dcid) to 5, and enable the regular sending of KeepAlive messages.
<Sysname> system-view [Sysname] interface serial 1/0 [Sysname-Serial1/0] link-protocol fr ietf [Sysname-Serial1/0] fr dlci 100 [Sysname-fr-dlci-Serial1/0-100] vofr nonstandard-compatible signal-channel 4 data-channel 5 keepalive

# Set the call control protocol on DLCI 200 to Huawei-compatible (DTE).

<Sysname> system-view [Sysname] interface serial 1/0 [Sysname-Serial1/0] link-protocol fr ietf [Sysname-Serial1/0] fr dlci 200 [Sysname-fr-dlci-Serial1/0-100] vofr huawei-compatible dte

vofr frf11-timer
Description
Use vofr frf1 1-timer to configure the trunk wait timer length in the FRF.1 trunk mode. 1 Use undo vofr frf1 1-timer to restore the default. By default, the trunk wait timer length is 30 seconds. This command has global significance. The configuration is valid for all FRF.1 trunk calls after the command 1 is executed. Related commands: call-mode.

292

NOTE:
The Trunk Wait timer is specific to the FRF.1 trunk mode. Within the trunk wait timer length, incoming calls are 1 prohibited and received voice packets are dropped. No signaling is exchanged in the FRF.1 trunk mode. When one voice gateway receives the first voice packet from 1 its peer voice gateway over a dedicated voice channel, the former considers that a call is coming. When either party involved in a call hangs up, the peer voice gateway (relative to the party who hangs up) will still keep sending voice packets to the local voice gateway. Without the Trunk Wait timer mechanism, the local voice gateway will immediately alert the party who has hung up so that this party could never hang up successfully in the FRF.1 trunk 1 mode.

Syntax
vofr frf1 1-timer time undo vofr frf1 1-timer

View
Voice view

Default level
2: System level

Parameters
time: Trunk Wait timer length in the FRF.1 trunk mode in seconds, in the range of 10 to 600. 1

Examples
# Configure the Trunk Wait timer length in the FRF.1 trunk mode to 40 seconds. 1
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] vofr frf11-timer 40

293

Voice RADIUS configuration commands

aaa-client
Description
Use aaa-client to enter voice AAA client view.

Syntax
aaa-client

View
Voice view

Default level
2: System level

Parameters
None

Examples
# Enter voice AAA client view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] aaa-client [Sysname-voice-aaa]

accounting
Description
Use accounting to enable the RADIUS accounting function for users who dial some access number. Use undo accounting to disable the RADIUS accounting function. By default, the RADIUS accounting function is disabled for users who dial access numbers. On one voice gateway, the RADIUS accounting function for one-stage dialing users (who dial a called number to originate a call after picking up the phone) differs from that for two-stage dialing users (who first dial an access number and then a called number to originate a call after picking up the phone). This command is only applicable to an access number, two-stage dialing users. With the RADIUS accounting function enabled, the RADIUS server will perform accounting for all users who use this access number. With the function disabled, the RADIUS server will not perform accounting for users who dial the access number. Related commands: gw-access-number, acct-method, and accounting-did.

Syntax
accounting undo accounting

294

View
Access number view

Default level
2: System level

Parameters
None

Examples
# Enable the RADIUS accounting function for users who dial the access number 17909.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] accounting

# Disable the RADIUS accounting function for users who dial the access number 17909.
[Sysname-voice-dial-anum17909] undo accounting

accounting-did
Description
Use accounting-did to enable the RADIUS accounting function for all one-stage dialing users. Use undo accounting-did to disable the RADIUS accounting function. By default, the RADIUS accounting function is disabled for all one-stage dialing users. On one voice gateway, the RADIUS accounting for one-stage dialing users is separated from that for two-stage dialing users. This command is applicable to only one-stage dialing users. With this function enabled, the RADIUS server will perform RADIUS accounting for all calls originated by one-stage dialing users. With this function disabled, the RADIUS server will not perform accounting for any calls originated by one-stage dialing users. Related commands: acct-method and accounting.

Syntax
accounting-did undo accounting-did

View
Voice AAA client view

Default level
2: System level

Parameters
None

Examples
# Enable the accounting function for all one-stage dialing users.
<Sysname> system-view

295

[Sysname] voice-setup [Sysname-voice] aaa-client [Sysname-voice-aaa] accounting-did

# Disable the accounting function for all one-stage dialing users.

[Sysname-voice-aaa] undo accounting-did

acct-method
Description
Use acct-method to configure an accounting method for the RADIUS client. Use undo acct-method to restore the default. By default, the accounting method is start-no-ack. Related commands: accounting and accounting-did.

Syntax
acct-method { start-ack | start-no-ack | stop-only } undo acct-method

View
Voice AAA client view

Default level
2: System level

Parameters
start-ack: When the call setup begins, the voice gateway sends an Accounting-Start request to the RADIUS server. However, the voice gateway must receive an Accounting_Start acknowledgment from the RADIUS server before connecting the call. After the call ends, the voice gateway sends an Accounting_Stop request to the RADIUS server, and releases the call upon receiving an Accounting_Stop acknowledgment. start-no-ack: When the call setup begins, the voice gateway sends an Accounting_Start request to the RADIUS server, and directly connects the call without waiting for an Accounting_Start acknowledgment. If the voice gateway receives an Accounting_Start unacknowledgment from the RADIUS server after the call is connected, it immediately releases the call. After the call ends, the voice gateway sends an Accounting_Stop request to the RADIUS server and releases the call only after it receives an Accounting_Stop acknowledgment. stop-only: The voice gateway sends an Accounting_Stop request to the RADIUS server only when the call ends, and it releases the call only after receiving an Accounting_Stop acknowledgment.

Examples
# Set the accounting method to start-ack.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] aaa-client [Sysname-voice-aaa] acct-method start-ack

# Restore the default accounting method.

[Sysname-voice-aaa] undo acct-method

296

authentication
Description
Use authentication to enable the RADIUS authentication function for users who dial some access number. Use undo authentication to disable the RADIUS authentication function. By default, the RADIUS authentication function is disabled for users who dial access numbers. For each access number, you can specify the RADIUS server to perform authentication for users who dial it. If the authentication function is enabled for users who dial some access number, only users who pass authentication can be authorized to make IP calls. If the authentication function is disabled, users who dial the access number can directly make IP calls no matter whether they are legal. The authentication function must be enabled before the authorization function. When the authentication function is disabled, the authorization function will automatically be disabled, and meanwhile, the authorization and undo authorization commands will be unavailable. Related commands: gw-access-number and authorization.

Syntax
authentication undo authentication

View
Access number view

Default level
2: System level

Parameters
None

Examples
# Enable the authentication function for users who dial the access number 17909.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] authentication

# Disable the authentication function for users who dial the access number 17909.
[Sysname-voice-dial-anum17909] undo authentication

authentication-did
Description
Use authentication-did to enable the authentication function for all one-stage dialing users. Use undo authentication-did to disable the authentication function. By default, the authentication function is disabled for all one-stage dialing users. This command is applicable to only one-stage dialing users, instead of two-stage dialing users.
297

With this function enabled, the calling number of one-stage dialing users who want to make IP calls is sent to the RADIUS server for authentication. Only users who pass authentication can make IP calls. Those who fail authentication will be disconnected and cannot make IP calls. The authentication function must be enabled before the authorization function. When the authentication function is disabled, the authorization function will automatically be disabled, and meanwhile, the authorization-did and undo authorization-did commands will be unavailable. Related commands: authorization-did.

Syntax
authentication-did undo authentication-did

View
Voice AAA client view

Default level
2: System level

Parameters
None

Examples
# Enable the authentication function for one-stage dialing users.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] aaa-client [Sysname-voice-aaa] authentication-did

authorization
Description
Use authorization to enable the authorization function for users who dial some access number. Use undo authorization to disable the authorization function. By default, the authorization function is disabled for users who dial access numbers. With this function enabled, called numbers will be sent to the RADIUS server for authorization after users who dial some access number to make IP calls pass authentication. You must enable the authentication function (by using the authentication command) before the authorization function. Otherwise, authorization is unavailable. Related commands: gw-access-number and authentication.

Syntax
authorization undo authorization

View
Access number view

298

Default level
2: System level

Parameters
None

Examples
# Enable the authorization function for users who dial the access number 17909.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] authentication [Sysname-voice-dial-anum17909] authorization

# Disable the authorization function for users who dial the access number 17909.
[Sysname-voice-dial-anum17909] undo authorization

authorization-did
Description
Use authorization-did to enable the authentication function for all one-stage dialing users. Use undo authorization-did to disable the authorization function for all one-stage dialing users. By default, the authorization function is disabled for all one-stage dialing users. This command is applicable to only one-stage dialing users, instead of two-stage dialing users. With this function enabled, called numbers will be sent to the RADIUS server for authorization after users who dial some access number to make IP calls pass authentication. You must enable the authentication function before the authorization function. Otherwise, authorization-did is unavailable. Related commands: authentication-did.

Syntax
authorization-did undo authorization-did

View
Voice AAA client view

Default level
2: System level

Parameters
None

Examples
# Enable the authorization function for one-stage dialing users.
<Sysname> system-view [Sysname] voice-setup

299

[Sysname-voice] aaa-client [Sysname-voice-aaa] authentication-did [Sysname-voice-aaa] authorization-did

# Disable the authorization function for one-stage dialing users.

[Sysname-voice-aaa] undo authorization-did

callednumber receive-method
Description
Use callednumber receive-method to configure the method of collecting digits of a called number. Use undo callednumber receive-method to restore the default. By default, users need to press the dial terminator # after dialing all digits of a called number. This command is applicable to both the one-stage dialing process and two-stage dialing process. In the terminator mode, the voice gateway can immediately originate a call only after users dial a called number and press the dial terminator #, and otherwise, the voice gateway will not originate a call until timeout. In the immediate mode, the voice gateway can originate a call immediately it collects all digits of a called number, without waiting for users to press the dial terminator #. The immediate mode simplifies users operations. Related commands: gw-access-number.

Syntax
callednumber receive-method { immediate | terminator } undo callednumber receive-method

View
Access number view

Default level
2: System level

Parameters
immediate: Specifies the voice gateway to originate a call immediately it collects all digits of a called number. terminator: Specifies users to press the dial terminator # after dialing a called number.

Examples
# Set the method of collecting digits of called numbers to immediate for the access number 17909.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] callednumber receive-method immediate

# Restore the default method of collecting digits of called numbers for the access number 17909.
[Sysname-voice-dial-anum17909] undo callednumber receive-method

300

card-digit
Description
Use card-digit to configure the number of digits in a card number for some access number in the card number/password process. Use undo card-digit to restore the default. By default, the number of digits in a card number is 12 only when an access number is already configured for the card number/password process (by using the process-config command). This command is used to configure the number of digits in a card number for the card number/password process. Once the number of digits is fixed, all users who use the access number must enter a fixed-length card number. Otherwise, the voice gateway will report an error. The card-digit command is available in access number view only after you use process-config to specify the dialing process as card number/password process. Related commands: gw-access-number and process-config.

Syntax
card-digit card-digit undo card-digit

View
Access number view

Default level
2: System level

Parameters
card-digit: Number of digits in a card number, in the range of 1 to 31.

Examples
# Specify the number of digits in a card number as 10 for the access number 17909.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] process-config cardnumber [Sysname-voice-dial-anum17909] card-digit 10

cdr
Description
Use cdr to configure a rule for saving CDRs. Use undo cdr to restore the default saving rule, and undo cdr all to restore the values of buffer, duration, and threshold all to the defaults. The voice gateway will save a certain amount of CDRs according to the configured rule. When you set the number of CDRs that can be saved or the lifetime of CDRs, the voice gateway will judge whether the existing CDRs will be deleted. If so, the voice gateway will prompt for confirmation and determine whether to validate the configuration according to your confirmation.
301

If both the buffer and duration keywords are specified, the number of saved CDRs cannot exceed the limit set by the buffer keyword. If large traffic is generated in a period of time, the CDRs for the calls completed earliest will be removed to keep the number of saved CDRs under the limit even if they have not reached the lifetime. Related commands: display voice call-history-record.

Syntax
cdr { buffer size-number | duration time-length | threshold percentage } undo cdr { all | buffer | duration | threshold }

View
Voice AAA client view

Default level
2: System level

Parameters
buffer size-number: Specifies the number of CDRs that can be saved in the buffer. The size-number argument ranges from 0 to 500, with a default of 50. The value 0 indicates that no CDR can be saved. duration time-length: Specifies the lifetime of CDRs in seconds. The time-length argument ranges from 0 to 2,147,483,647, with a default of 86,400. The value 0 indicates that no CDR can be saved. threshold percentage: Specifies the alarm threshold in percentage for CDRs. When the percentage of the saved CDRs in the total CDRs that can be saved in the buffer reaches the alarm threshold, the voice gateway will generate alarm information once. The percentage argument ranges from 0 to 100, with a default of 80. The value 0 indicates that no alarm information will be output.

Examples
# Set the number of CDRs that can be saved to 400.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] aaa-client [Sysname-voice-aaa] cdr buffer 400

# Set the lifetime of CDRs to 10 hours.

[Sysname-voice-aaa] cdr duration 36000

# Set the alarm threshold for CDRs to 10%.

[Sysname-voice-aaa] cdr threshold 10

display voice access-number

Description
Use display voice access-number to display the configuration information and access numbers in voice AAA client view. The information displayed includes: Accounting method Enabling or disabling of the authentication, authorization, and accounting functions for one-stage dialing users Rule for saving CDRs
302

Configuration information for all access numbers

Related commands: gw-access-number and aaa-client.

Syntax
display voice access-number [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display the configuration information and access numbers.
<Sysname> display voice access-number AAA configuration : accounting-method accounting-did authentication-did authorization-did call history rule: cdr buffer cdr duration cdr threshold access number: [ 17909 ] dialing process accounting authentication authorization callednum receive card digit password digit redialing times access number: [ 201 ] dialing process accounting authentication authorization = = off = voice-caller = off off = = = = = on termintor = 6 2 12 = cardnumber = on on = = 50 = 86400 100 = = = start-ack = off on on

303

callednum receive redialing times = 2 = language selected

immediate

Chinese

Table 58 Output description Field

accounting-method

Description
Accounting method, including start-ack, start-no-ack, and stop-only. See the acct-method command. Accounting function for one-stage dialing users.

accounting-did

on: Enabled. off: Disabled.

See the accounting-did command. Authentication function for one-stage dialing users.

authentication-did

on: Enabled. off: Disabled.

See the authentication-did command. Authorization function for one-stage dialing users.

authorization-did

on: Enabled. off: Disabled.

See the authorization-did command. Rule for saving CDRs. Number of CDRs that can be saved. See the cdr buffer command. Lifetime of CDRs. See the cdr duration command. CDR alarm threshold. See the cdr threshold command. Access number, for example, 17909. See the gw-access-number command. Two-stage dialing process, including card number/password process, caller number process, caller number process with IVR. See the process-config command. Accounting function for two-stage dialing users.

call history rule cdr buffer cdr duration cdr threshold

access number

dialing process

accounting

on: Enabled. off: Disabled.

See the accounting command. Authentication function for two-stage dialing users.

authentication

on: Enabled. off: Disabled.

See the authentication command.

304

Field

Description
Authorization function for two-stage dialing users

authorization

on: Enabled off: Disabled

See the authorization command. Method of collecting digits of a called number, including terminator and immediate See the callednumber receive-method command.

callednum receive

card digit

Number of digits in a card number, displayed only in the card number/password process See card-digit command.

password digit

Number of digits in a password, displayed only in the card number/password process See the password-digit command.

redialing times

Number of redial attempts, displayed in the card number/password process or caller number process with IVR See the redialtimes command.

language selected

Language selection function, Chinese and English available, displayed only in the caller number process with IVR See the selectlanguage command.

display voice call-history-record

Description
Use display voice call-history-record to display voice RADIUS call records. If the ip-address argument is specified, the voice gateway displays call records by callees IP address. If the last-number argument is specified, the voice gateway displays the specified number of latest call records, and if a value greater than the number of actual call records is specified, the voice gateway will display all call records. The voice gateway finds call records by the search condition. If the voice gateway fails to find a call record or the found record is null, the voice gateway will give prompt information. Related commands: cdr.

Syntax
display voice call-history-record { all | callednumber called-number | callingnumber calling-number | cardnumber card-number | last last-number | line line-number | remote-ip-addr ip-address } [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

305

Parameters
all: Displays all call records. callednumber called-number: Displays call records by called number. The called-number argument is a string of up to 31 characters, consisting of digits 0 through 9 and the asterisk *. callingnumber calling-number: Displays call records by calling number. The calling-number argument is a string of up to 31 characters, consisting of digits 0 through 9 and the asterisk *. card card-number: Displays call records by prepaid card number. The card-number argument is a string of up to 31 characters. last last-number: Displays the specified number of latest call records. The last-number argument ranges from 1 to 500. line line-number: Displays incoming or outgoing call records by voice subscriber line of the voice gateway. remote-ip-addr ip-address: Displays call records by callees IP address. The ip-address argument represents a callees IP address. |: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display call records by calling number.
<Sysname> display voice call-history-record callingnumber 4000 Call records of voice RADIUS: # CallRecord [ 0 ]: CallReference CallRecordTime CardNumber AccessNumber = 46 = Oct 20, 2006 16:45:47 = None = None

Incoming call leg: CallingNumber SignalType SetupTime ConnectTime ReleaseTime SendPackets SendBytes ReceiveBytes = 4000 = FXS/O = Oct 20, 2006 16:45:43 = Oct 20, 2006 16:45:45 = Oct 20, 2006 16:45:47 = 71 packages = 2982 bytes = 4662 bytes

VoiceInterface = 1/0

ReceivePackets = 111 packages

Outgoing call leg [ 0 ]:

306

CalledNumber CallDuration EncodeType DecodeType ReleaseCause SignalType SetupTime ConnectTime ReleaseTime SendPackets SendBytes ReceiveBytes # The end

= 2000 = 00h 00m 02s = G729R8 = G729R8 = Called hook on = SIP = Oct 20, 2006 16:45:43 = Oct 20, 2006 16:45:45 = Oct 20, 2006 16:45:47 = 111 packages = 4662 bytes = 3024 bytes

IpAddress/Port = 1.1.1.19/5060

ReceivePackets = 72 packages

Table 59 Output description Field

Call records of voice RADIUS CallRecord [ 0 ] CallReference CallRecordTime CardNumber AccessNumber Incoming Call Leg CallingNumber SignalType VoiceInterface SetupTime ConnectTime ReleaseTime SendPackets SendBytes ReceivePackets ReceiveBytes Outgoing call leg [ 0 ] CalledNumber CallDuration EncodeType

Description
Voice RADIUS call records Call record number Voice RADIUS module call identification Time when a call is recorded Card number Access number Information of the incoming call leg Calling number Signaling protocol type (for example, R2, E&M) Voice interface Call setup time Call-connected time Call release time Packets sent Bytes sent Packets received Bytes received Information of the outgoing call leg. One call may involves multiple outgoing call legs. [ 0 ] identifies one outgoing call leg. Called number Call duration Encoding type 307

Field
DecodeType ReleaseCause SignalType VoiceInterface IpAddress/Port SetupTime ConnectTime ReleaseTime SendPackets SendBytes ReceivePackets ReceiveBytes

Description
Decoding type Call release cause Signaling protocol (for example, R2, E&M) Voice interface IP address and port number Call setup time Call-connected time Call release time Packets sent Bytes sent Packets received Bytes received

display voice radius statistic

Description
Use display voice radius statistic to display statistics of messages exchanged between the voice RADIUS module, call management center (CMC) module, and AAA module. Related commands: reset voice radius statistic.

Syntax
display voice radius statistic [ | { begin | exclude | include } regular-expression ]

View
Any view

Default level
2: System level

Parameters
|: Filters command output by specifying a regular expression. For more information about regular expressions, see Fundamentals Configuration Guide. begin: Displays the first line that matches the specified regular expression and all lines that follow. exclude: Displays all lines that do not match the specified regular expression. include: Displays all lines that match the specified regular expression. regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.

Examples
# Display statistics of messages exchanged between the voice RADIUS module, CMC module, and AAA module.
<Sysname> display voice radius statistic VORDS => AAA:

308

Authen_Request Author_Request AcctReq_PstnCaller AcctReq_VoipCaller AcctReq_PstnCalled AcctReq_VoipCalled Account_Stop Leaving AAA => VORDS: Authen_Accept Authen_Reject Author_Accept Author_Reject AcctRsp_PstnCaller AcctRsp_VoipCaller AcctRsp_PstnCalled AcctRsp_VoipCalled Account_Ok Account_Failure Cut CMC => VORDS: Setup Alerting Connect Release DtmfInformation ChannelReady FaxVoiceSwitch FaxTone = = = 0 0 0 = = = = 0 0 0 0 = = = = = = 0 = 0 = = = 0 0 0 0 0 0 0 0 = = = = 0 0 0 0

= =

0 0

0 = = 0 0

Table 60 Output description Field

VORDS=>AAA: Authen_Request Author_Request AcctReq_PstnCaller AcctReq_VoipCaller AcctReq_PstnCalled AcctReq_VoipCalled Account_Stop Leaving AAA=>VORDS: Authen_Accept Authen_Reject

Description
Messages from the voice RADIUS module to the AAA module Authentication_Request message Authorization_Request message Accounting_Request message for PSTN caller Accounting_Request message for VoIP caller Accounting_Request message for PSTN callee Accounting_Request message for VoIP callee Accounting_Stop message Leaving message Messages from the AAA module to the voice RADIUS module Authentication_Accept message Authentication_Reject message 309

Field
Author_Accept Author_Reject AcctRsp_PstnCaller AcctRsp_VoipCaller AcctRsp_PstnCalled AcctRsp_VoipCalled Account_Ok Account_Failure Cut CMC=>VORDS: Setup Alerting Connect Release DtmfInformation ChannelReady FaxVoiceSwitch FaxTone

Description
Authorization_Accept message Authorization_Reject message Accounting_Response message for PSTN caller Accounting_Response message for VoIP caller Accounting_Response message for PSTN callee Accounting_Response message for VoIP callee Accounting_Ok message Accounting_Failure message Cut message Messages from the CMC module to the voice RADIUS module Setup message Alerting message Connect message Release message DTMF digit Channel_Ready message Fax_Voice_Switch message Fax_Tone message

gw-access-number
Description
Use gw-access-number to configure an access number or enter access number view. Use undo gw-access-number to delete one or all access numbers. By default, no access number is configured. When you delete all configured access numbers, the voice gateway will give alarm information, requiring you to make a confirmation. You can press <Y> to delete all access numbers or press <N> to cancel the operation. An access number can contain up to 31 characters, but no unacceptable characters such as a letter. At most 100 access numbers can be configured for the voice gateway. The shortest match and exact match are preferred for access number match. If an access number template is the same as a voice entity template, the global number substitution rules in voice dial program view and those in voice subscriber line view will be valid for the access number, but no entity substitution rule can be matched in access number view.

Syntax
gw-access-number access-number undo gw-access-number { access-number | all }
310

View
Voice dial program view

Default level
2: System level

Parameters
access-number: Access number (for example, 169 and 17909), a string of up to 31 characters consisting of digits 0 through 9 and the wildcard .. The wildcard . represents a digital character and must follow a digit or appear separately. all: Deletes all access numbers.

Examples
# Add the access number 17909 and enter access number view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909]

# Add the access number 179 and enter access number view.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 179.. [Sysname-voice-dial-anum179..]

# Delete the access number 17909.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] undo gw-access-number 17909

# Delete all access numbers.

<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] undo gw-access-number all Delete all access numbers, are you sure? (Y/N) y

password-digit
Description
Use password-digit to configure the number of digits in a password for some access number in the card number/password process. Use undo password-digit to restore the default number of digits in a password for some access number in the card number/password process. This command is unavailable for the caller number process with IVR. By default, the number of digits in a password for some access number in the card number/password process is 6.
311

Before executing the password-digit command, you must use process-config to specify the two-stage dialing process for the configured access number as card number/password process. The password-digit command is available only in access number view. Related commands: gw-access-number and process-config.

Syntax
password-digit password-digit undo password-digit

View
Access number view

Default level
2: System level

Parameters
password-digit: Number of digits in a password, in the range of 1 to 16.

Examples
# Specify the number of digits in a password as 4 for the access number 17909.
<Sysname>system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] process-config cardnumber [Sysname-voice-dial-anum17909] password-digit 4

# Restore the default number of digits in a password for the access number 17909.
[Sysname-voice-dial-anum17909] undo password-digit

process-config
Description
Use process-config to specify a dialing process for an access number. Use undo process-config to restore the default dialing process for an access number. By default, the caller number process with IVR is specified for all access numbers. Each access number has a specific dialing process. Calls originated by users who dial a certain access number are established in accordance with the same dialing process. The caller number process and the caller number process with IVR differ in two ways: In the caller number process, after a user dials an access number, the voice gateway plays only dial tones (long tones). In the caller number process with IVR, after a user dials an access number, the voice gateway will play prompt tones, requiring the user to dial a called number.

In the card number/password process, with the authentication function disabled, a user can enter any two numbers as a card number and password respectively to make an IP call as long as they meet the length requirements.

312

After a dialing process is specified, parameters not related to the process are set to the default values and the corresponding commands are unavailable. Parameters related to the card number/password process include number of digits in a card number and number of digits in a password. The language selection function is applicable only to the caller number process with IVR, while the number of redial attempts is applicable to only the card number/password process and the caller number process with IVR. Related commands: gw-access-number, card-digit, password-digit, and selectlanguage.

Syntax
process-config { callernumber | cardnumber | voice-caller } undo process-config

View
Access number view

Default level
2: System level

Parameters
callernumber: Specifies the two stage-dialing process as caller number process. After a user dials an access number, the voice gateway will continue to play dial tones, prompting for a called number. In this process, the user authentication is implemented by identifying the calling number, and no more additional parameter configurations are required. cardnumber: Specifies the two-stage dialing process as card number/password process. After a user dials an access number, the voice gateway will continue to play prompt tones, requiring the user to enter a card number and password. In this process, the user authentication is implemented by identifying the prepaid card number and password, and you can configure parameters by using the card-digit, password-digit, and redialtimes commands. voice-caller: Specifies the two-stage dialing process as caller number process with IVR. After a user dials an access number, the voice gateway will play prompt tones, requiring the user to dial a called number. In this process, the user authentication is implemented by identifying the calling number. If the authentication succeeds, the voice gateway plays prompt tones, requiring the user to dial a called number. In addition, you can configure the number of redial attempts by using the redialtimes command, and the language in which the prompt tones are played by using the selectlanguage command.

Examples
# Specify the dialing process for the access number 17909 as card number/password process.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] process-config cardnumber

# Restore the default dialing process for the access number 17909.
[Sysname-voice-dial-anum17909] undo process-config

redialtimes
Description
Use redialtimes to configure the number of redial attempts in each dialing step for an access number.
313

Use undo redialtimes to restore the default number of redial attempts for an access number. By default, the number of redial attempts in each dialing step is 2 for an access number. The redialtimes-number argument refers to the number of redial attempts, that is, the number of dial attempts is the number of redial attempts plus 1. This command is unavailable in the caller number process. For the card number/password process, you can use redialtimes to set times of reselecting a language and times of redialing a card number, password, or called number. To make an IP call, a user first dials an access number, then selects a language, next enters a prepaid card number and password, and finally dials a called number. Any error in each dialing step may lead to a dialing failure. For the caller number process with IVR, you can use redialtimes to set times of reselecting a language and times of redialing a called number. Related commands: gw-access-number and process-config.

Syntax
redialtimes redialtimes-number undo redialtimes

View
Access number view

Default level
2: System level

Parameters
redialtimes-number: Number of redial attempts, in the range of 0 to 10. In the card number/password process, this argument may refer to the times of reselecting a language or redialing a card number, password, or a called number. In the caller number process with IVR, this argument may refer to the times of reselecting a language or redialing a called number.

Examples
# Set the number of redial attempts to 4 for the access number 17909.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] process-config cardnumber [Sysname-voice-dial-anum17909] redialtimes 4

reset voice radius statistic

Description
Use reset voice radius statistic to clear statistics of messages exchanged between the voice RADIUS module, CMC module, and AAA module. Related commands: display voice radius statistic.

Syntax
reset voice radius statistic
314

View
User view

Default level
2: System level

Parameters
None

Examples
# Clear the statistics of messages exchanged between the voice RADIUS module, CMC module, and AAA module.
<Sysname> reset voice radius statistic

selectlanguage
Description
Use selectlanguage to configure a language in which prompt tones are played in the caller number process with IVR. Use undo selectlanguage to restore the default. By default, prompt tones are played in Chinese. This command is available only in the caller number process with IVR. Related commands: gw-access-number and process-config.

Syntax
selectlanguage { enable | chinese | english } undo selectlanguage

View
Access number view

Default level
2: System level

Parameters
enable: Enables the language selection function so that users can select a language to play prompt tones. chinese: Plays prompt tones in Chinese. english: Plays prompt tones in English.

Examples
# Configure the voice gateway to play prompt tones in English.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] process-config voice-caller [Sysname-voice-dial-anum17909] selectlanguage english

315

timer two-stage dial-interval

Description
Use timer two-stage dial-interval to configure the timeout interval for a user to dial the next digit in a two-stage dialing process. Use undo timer two-stage dial-interval to restore the default. By default, the timeout interval is 10 seconds. A timer resets every time the user dials a digit until all the digits are dialed. If the timer times out before the dialing finishes, there are two scenarios: In the card number/password process and caller number process with IVR, if the number of redial attempts is not reached, the user is prompted to redial the number In the caller number process, or if the number of redial attempts is reached, the user is prompted to hang up, and the call ends.

Syntax
timer two-stage dial-interval seconds undo timer two-stage dial-interval

View
Access number view

Default level
2: System level

Parameters
seconds: Timeout interval between two digits in a two-stage dialing process, ranging from 1 to 300, in seconds.

Examples
# Configure the timeout interval between two digits as 5 seconds for users who dial the access number 17909.
<Sysname> system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] gw-access-number 17909 [Sysname-voice-dial-anum17909] timer two-stage dial-interval 5

316

Support and other resources

Contacting HP
For worldwide technical support information, see the HP support website: http://www.hp.com/support Before contacting HP, collect the following information: Product model names and numbers Technical support registration number (if applicable) Product serial numbers Error messages Operating system type and revision level Detailed questions

Subscription service
HP recommends that you register your product at the Subscriber's Choice for Business website: http://www.hp.com/go/wwalerts After registering, you will receive email notification of product enhancements, new driver versions, firmware updates, and other product resources.

Related information
Documents
To find related documents, browse to the Manuals page of the HP Business Support Center website: http://www.hp.com/support/manuals For related documentation, navigate to the Networking section, and select a networking category. For a complete list of acronyms and their definitions, see HP A-Series Acronyms.

Websites
HP.com http://www.hp.com HP Networking http://www.hp.com/go/networking HP manuals http://www.hp.com/support/manuals HP download drivers and software http://www.hp.com/support/downloads HP software depot http://www.software.hp.com

317

Conventions
This section describes the conventions used in this documentation set.

Command conventions
Convention
Boldface Italic [] { x | y | ... } [ x | y | ... ] { x | y | ... } * [ x | y | ... ] * &<1-n> #

Description
Bold text represents commands and keywords that you enter literally as shown. Italic text represents arguments that you replace with actual values. Square brackets enclose syntax choices (keywords or arguments) that are optional. Braces enclose a set of required syntax choices separated by vertical bars, from which you select one. Square brackets enclose a set of optional syntax choices separated by vertical bars, from which you select one or none. Asterisk-marked braces enclose a set of required syntax choices separated by vertical bars, from which you select at least one. Asterisk-marked square brackets enclose optional syntax choices separated by vertical bars, from which you select one choice, multiple choices, or none. The argument or keyword and argument combination before the ampersand (&) sign can be entered 1 to n times. A line that starts with a pound (#) sign is comments.

GUI conventions
Convention
Boldface >

Description
Window names, button names, field names, and menu items are in bold text. For example, the New User window appears; click OK. Multi-level menus are separated by angle brackets. For example, File > Create > Folder.

Symbols
Convention
WARNING CAUTION IMPORTANT NOTE TIP

Description
An alert that calls attention to important information that if not understood or followed can result in personal injury. An alert that calls attention to important information that if not understood or followed can result in data loss, data corruption, or damage to hardware or software. An alert that calls attention to essential information. An alert that contains additional or supplementary information. An alert that provides helpful information.

318

Network topology icons

Represents a generic network device, such as a router, switch, or firewall. Represents a routing-capable device, such as a router or Layer 3 switch. Represents a generic switch, such as a Layer 2 or Layer 3 switch, or a router that supports Layer 2 forwarding and other Layer 2 features.

Port numbering in examples

The port numbers in this document are for illustration only and might be unavailable on your device.

319

Index
ABCDEFGHIJKLMNOPQRSTUVW
A aaa-client,294 account enable,213 accounting,294 accounting-did,295 acct-method,296 address,21 1 address,284 address sip,155 address sip server-group,212 amd enable,93 amd parameter,93 Analog voice subscriber line configuration commands,45 ani,94 ani-offset,95 answer enable,96 area,45 area-prefix,197 assign,212 authentication,297 authentication,197 authentication-did,297 authorization,298 authorization-did,299 B backup-rule loose,229 bind sip-trunk account,214 busytone-hookon timer,46 busytone-t-th,46 C callednumber receive-method,300 caller-group,129 caller-permit,129 call-fallback,156 call-forwarding no-reply enable,229
320

call-forwarding on-busy enable,230 call-forwarding priority,231 call-forwarding unavailable enable,231 call-forwarding unconditional enable,232 call-history,1 call-hold enable,233 call-hold-format,233 calling-name,47 callmode,96 call-mode,285 call-normal,267 call-route,198 call-rule-set,199 call-transfer enable,234 call-transfer start-delay,235 call-waiting,235 call-waiting enable,236 call-waiting priority,237 call-watch group,249 call-watch rule,250 card-digit,301 cas,97 cdr,301 cid display,48 cid receive,48 cid ring,49 cid select-mode,285 cid send,50 cid type,50 clear-forward-ack enable,98 cng-on,51 codec transparent,215 compression,1 conference enable,237 cptone country-type,52 cptone tone-type,54 crypto,156

D default,55 default entity compression,7 default entity fax,253 default entity payload-size,8 default entity vad-on,9 default subscriber-line,56 delay hold,56 delay rising,57 delay send-dtmf,58 delay send-wink,58 delay start-dial,60 delay wink-hold,59 delay wink-rising,59 description,215 description,268 description,131 description (voice entity view),10 description (voice subscriber line view),61 dialin-restriction enable,238 dialout-restriction enable,239 dial-prefix,132 dial-program,1 1 dial-trap enable,10 Digital voice subscriber line configuration commands,93 disconnect lcfo,61 display call-watch status,251 display fr vofr-info,286 display voice access-number,302 display voice call-history-record,305 display voice call-info,1 1 display voice cmc,13 display voice default all,16 display voice entity,17 display voice enum-group,161 display voice fax,255 display voice ipp statistic,19 display voice iva statistic,21 display voice ivr call-info,268 display voice ivr media-play,269 display voice ivr media-source,270 display voice number-substitute,134 display voice radius statistic,308 display voice server-group,217
321

display voice sip call-statistics,157 display voice sip connection,160 display voice sip dns-record,162 display voice sip reason-mapping,162 display voice sip register-state,166 display voice sip subscribe-state,239 display voice sip-server register-user,200 display voice sip-server resource-statistic,201 display voice sip-trunk account,216 display voice ss mwi,240 display voice statistics call-active,22 display voice statistics call-history,25 display voice statistics entity,28 display voice subscriber-group,133 display voice subscriber-line,62 display voice subscriber-line,99 distinguish-localtalk,30 dl-bits,100 dns-type,165 dot-match,135 dscp media,30 dtmf amplitude,65 dtmf enable,102 dtmf sensitivity-level,65 dtmf threshold,67 dtmf threshold digital,102 dtmf time,66 E early-media enable,167 echo-canceller,69 echo-canceller parameter,70 em-passthrough,72 em-phy-parm,71 em-signal,71 enable snmp trap updown,103 entity,31 entity ivr,271 entity vofr,287 enum-group,168 expires,202 extension,272 F fax baudrate,258

fax cng-switch enable,259 fax ecm,259 fax level,260 fax local-train threshold,261 fax nsf-on,261 fax protocol,262 fax train-mode,263 feature,242 final-callednum enable,104 first-rule,136 force-metering enable,104 G group-b enable,105 group-name,218 gw-access-number,310 H hookoff-mode,72 hookoff-mode delay bind,73 hookoff-time,74 hot-swap enable,219 hunt-group enable,243 hunt-group priority,243 I impedance,74 input-error,273 ivr-input-error,274 ivr-root,275 ivr-system,275 ivr-timeout,276 J joined-conference enable,244 K keepalive,168 keepalive,219 L line,106 line,32 line-check enable,169 listen transport,170 M match destination host-prefix,221
322

match source address,222 match source host-prefix,220 match-template,136 match-template,32 max-call (voice dial program view),138 max-call (voice entity view),139 media-file,277 media-play,277 media-protocol,171 mode,106 mode,203 modem compatible-param,264 modem protocol,265 mwi enable,245 mwi tone-duration,245 mwi-server,246 N nlp-on,75 node,278 number,203 number-match,139 number-priority,140 number-substitute,141 O open-trunk,76 operation,279 outband,35 outband sip,171 outband vofr,288 outbound-proxy,172 P password-digit,31 1 payload-size,35 pcm,108 plc-mode,77 posa called-length,108 priority,141 pri-set,109 privacy,173 private-line,142 probe remote-server,204 process-config,312 proxy,173

proxy server-group,223 Q qsig-tunnel enable,1 10 R re-answer enable,1 10 reason-mapping pstn,174 reason-mapping sip,176 receive gain,78 redialtimes,313 redundancy mode,179 redundancy mode,225 register enable,224 register-enable,178 register-number,36 register-user,205 register-value,1 1 1 registrar,179 registrar server-group,223 remote-party-id,181 renew,1 13 reset voice cmc statistic,37 reset voice cmc statistic,78 reset voice fax statistics,265 reset voice ipp statistic,37 reset voice ipp statistic,79 reset voice iva statistic,79 reset voice iva statistic,38 reset voice radius statistic,314 reset voice sip connection,181 reset voice sip dns-record,182 reset voice sip statistics,182 reverse,1 14 ring-detect debounce,80 ring-detect frequency,81 rtp payload-type nte,38 rule,183 rule,143 rule,205 S seizure-ack enable,1 15 selectlanguage,315 select-mode,1 15 select-rule operation-order,280
323

select-rule rule-order,147 select-rule search-stop,148 select-rule type-first,149 select-stop,150 send-busytone,81 send-number,150 send-ring,39 sendring ringbusy enable,1 16 seq-number,288 server enable,207 server-bind ipv4,207 server-group,225 service,206 set-media,280 shutdown (voice entity view),40 shutdown (voice subscriber line view),82 signal-value,1 17 silence-th-span,83 sip,183 sip-comp,184 sip-comp agent,185 sip-comp server,186 sip-domain,186 sip-server,208 sip-trunk account,226 sip-trunk enable,227 slic-gain,83 source-bind,187 special-character,1 18 srs,199 subscriber-group,151 subscriber-line,84 subscriber-line,1 19 substitute (voice dial program view),153 substitute (voice subscriber line view, voice entity view),152 T tdm-clock,1 19 terminator,154 timeout,281 timer called-hookon-delay,247 timer connection age,188 timer dial-interval,84 timer disconnect-pulse,85

timer dl,120 timer dtmf,121 timer first-dial,85 timer hold,122 timer hookflash-detect,86 timer hookoff-interval,87 timer register-complete group-b,124 timer register-pulse persistence,123 timer registration divider,189 timer registration expires,189 timer registration retry,188 timer registration threshold,190 timer ring,124 timer ring-back,87 timer session-expires,191 timer two-stage dial-interval,316 timer wait-digit,88 timeslot-set,125 timestamp,289 transmit gain,88 transport,191 trunk,209 trunk-direction,126 trunk-id,290 trusted-point,209 ts,127 type,89 U uri,192 url,193 user,194 user,227 user-input,282 V vad-on,40 vi-card busy-tone-detect,90 vi-card cptone-custom,91 vi-card reboot,92 vofr,291 vofr frf1 1-timer,292 voice bandwidth,290 voice-setup,41 voip timer,42
324

vqa dscp,42 vqa dsp-monitor buffer-time,44 W wildcard-register enable,195