HP a-MSR Router Series High Voice Command Reference

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 2011 Hewlett-Packard Development Company, L.P.

No part of this documentation may be reproduced or transmitted in any form or by any means without prior

written consent of Hewlett-Packard Development Company, L.P.

The information contained herein is subject to change without notice.

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.

iii

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

iv

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

v

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

vi

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

vii

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

viii

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

ix

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

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Index ············································································································································································· 320

1

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 g711alaw, that with the

third priority is g711ulaw, and that with the fourth priority is g723r53.

g711alaw and g711ulaw 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.

2

Table 1 Relationship between algorithms and bandwidth

Codec Bandwidth Voice quality

G.711 (A-law and µ-law) 64 kbps (without compression) Best

G.726 16, 24, 32, 40 kbps Good

G.729 8 kbps Good

G.723 r63 6.3 kbps Fair

G.723 r53 5.3 kbps Fair

Actual network bandwidth is related to packet assembly interval and network structure. The longer the packet

assembly interval is, the closer the network bandwidth is to the media stream bandwidth 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 µ-law)

Packet assembly interval

Bytes coded in a time unit

Packet length (IP) (bytes)

Network bandwidth (IP)

Packet length (IP+PPP) (bytes)

Network bandwidth (IP+PPP)

Coding latency

10 ms 80 120 96 kbps 126 100.8

kbps 10 ms

20 ms 160 200 80 kbps 206 82.4 kbps 20 ms

30 ms 240 280 74.7 kbps 286 76.3 kbps 30 ms

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

ms.

Table 3 G.723 r63 algorithm







Coding latency

30 ms 24 64 16.8 kbps 70 18.4 kbps 30 ms

60 ms 48 88 11.6 kbps 94 12.3 kbps 60 ms

90 ms 72 112 9.8 kbps 118 10.3 kbps 90 ms

120 ms 96 136 9.1 kbps 142 9.5 kbps 120 ms

150 ms 120 160 8.5 kbps 166 8.9 kbps 150 ms

180 ms 144 184 8.2 kbps 190 8.4 kbps 180 ms

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

3








Coding latency

30 ms 20 60 15.9 kbps 66 17.5 kbps 30 ms

60 ms 40 80 10.6 kbps 86 11.4 kbps 60 ms

90 ms 60 100 8.8 kbps 106 9.3 kbps 90 ms

120 ms 80 120 8 kbps 126 8.4 kbps 120 ms

150 ms 100 140 7.5 kbps 146 7.8 kbps 150 ms

180 ms 120 160 7.1 kbps 166 7.4 kbps 180 ms

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








Coding latency

10 ms 20 60 48 kbps 66 52.8 kbps 10 ms

20 ms 40 80 32 kbps 86 34.4 kbps 20 ms

30 ms 60 100 26.7 kbps 106 28.3 kbps 30 ms

40 ms 80 120 24 kbps 126 25.2 kbps 40 ms

50 ms 100 140 22.4 kbps 146 22.1 kbps 50 ms

60 ms 120 160 21.3 kbps 166 11.4 kbps 60 ms

70 ms 140 180 20.6 kbps 186 21.3 kbps 70 ms

80 ms 160 200 20 kbps 206 20.6 kbps 80 ms

90 ms 180 220 19.5 kbps 226 20.1 kbps 90 ms

100 ms 200 240 19.2 kbps 246 19.7 kbps 100 ms

110 ms 220 260 18.9 kbps 266 19.3 kbps 110 ms

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








Coding latency

10 ms 30 70 56 kbps 76 60.8 kbps 10 ms

20 ms 60 100 40 kbps 106 42.4 kbps 20 ms

30 ms 90 130 34.7 kbps 136 36.3 kbps 30 ms

40 ms 120 160 32 kbps 166 33.2 kbps 40 ms

50 ms 150 190 30.4 kbps 196 31.2 kbps 50 ms

60 ms 180 220 29.3 kbps 226 30.1 kbps 60 ms

4







Coding latency

70 ms 210 250 28.6 kbps 256 29.3 kbps 70 ms






Network bandwidth IP



Coding latency

10 ms 40 80 64 kbps 86 68.8 kbps 10 ms

20 ms 80 120 48 kbps 126 50.4 kbps 20 ms

30 ms 120 160 42.7 kbps 166 44.3 kbps 30 ms

40 ms 160 200 40 kbps 206 41.2 kbps 40 ms

50 ms 200 240 38.4 kbps 246 39.4 kbps 50 ms









Coding latency

10 ms 50 90 72 kbps 96 76.8 kbps 10 ms

20 ms 100 140 56 kbps 146 58.4 kbps 20 ms

30 ms 150 190 50.7 kbps 196 52.3 kbps 30 ms

40 ms 200 240 48 kbps 246 49.2 kbps 40 ms


Table 9 G.729 algorithm







Coding latency

10 ms 10 50 40 kbps 56 44.8 kbps 10 ms

20 ms 20 60 24 kbps 66 26.4 kbps 20 ms

30 ms 30 70 18.7 kbps 76 20.3 kbps 30 ms

5







Coding latency

40 ms 40 80 16 kbps 86 17.2 kbps 40 ms

50 ms 50 90 14.4 kbps 96 15.4 kbps 50 ms

60 ms 60 100 13.3 kbps 106 14.1 kbps 60 ms

70 ms 70 110 12.6 kbps 116 13.3 kbps 70 ms

80 ms 80 120 12 kbps 126 12.6 kbps 80 ms

90 ms 90 130 11.6 kbps 136 12.1 kbps 90 ms

100 ms 100 140 11.2 kbps 146 11.7 kbps 100 ms

110 ms 110 150 10.9 kbps 156 11.3 kbps 110 ms

120 ms 120 160 10.7 kbps 166 11.1 kbps 120 ms

130 ms 130 170 10.5 kbps 176 10.8 kbps 130 ms

140 ms 140 180 10.3 kbps 186 10.6 kbps 140 ms

150 ms 150 190 10.1 kbps 196 10.5 kbps 150 ms

160 ms 160 200 10 kbps 206 10.3 kbps 160 ms

170 ms 170 210 9.9 kbps 216 10.2 kbps 170 ms

180 ms 180 220 9.8 kbps 226 10 kbps 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

Codec


Before compression After IPHC+PPP compression





G.729 30 76 20.3 kbps 34 9.1 kbps

G.723r63 24 70 18.4 kbps 28 7.4 kbps

G.723r53 20 66 17.5 kbps 24 6.4 kbps

G.726r16 60 106 28.3 kbps 64 17.1 kbps

6

Codec


Before compression After IPHC+PPP compression





G.726r24 90 136 36.3 kbps 94 25.1 kbps

G.726r32 120 166 44.3 kbps 124 33.1 kbps

G.726r40 150 196 52.3 kbps 154 41.1 kbps

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: g711alaw, g711ulaw, g723r53, and g729r8. By default, the

codec with the first priority is g729r8, the codec with the second priority is g711alaw, the codec with the third

priority is g711ulaw, 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 } { g711alaw | g711ulaw | 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).

g711alaw: G.711 A-law codec (defining the pulse code modulation technology), requiring a bandwidth of

64 kbps, usually adopted in Europe.

g711ulaw: G.711μ-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.


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.

7



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-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 g711alaw, the

codec with the third priority is g711ulaw, 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 } { g711alaw | g711ulaw | 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.

8

4th-level: Specifies a codec with the fourth priority (the lowest priority).

g711alaw: G.711 A-law codec (defining the pulse code modulation technology), requiring a bandwidth of

64 kbps, usually adopted in Europe.

g711ulaw: G.711μ-law codec, requiring a bandwidth of 64 kbps, usually adopted in North America and

Japan.



g726r16: G.726 Annex A codec. It uses the ADPCM technology, requiring a bandwidth of 16 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-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 { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } time-length

undo default entity payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 }

View


Default level

2: System level

Parameters

g711: Specifies the packetization period for g711 codec. It can be 10, 20 (the default), or 30 milliseconds.

9

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 110 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.711 codec to 30 milliseconds.




[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. 711 does not support VAD).

Related commands: vad-on.

Syntax

default entity vad-on

undo default entity vad-on

View


Default level

2: System level

Parameters

None

Examples

# Enable VAD globally.

10




[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


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-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

11

Default level

2: System level

Parameters

None

Examples

# Enable the trap function for VoIP voice entity 10.





[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




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

12

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 : 0x002C0060

Module ID : LGS CMC

#

End

# 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 : From CS

ViIfIndex : 0x002C00F0

Related module ==>

Module ID : LGS

Reference Numbers : 1

Module ID : CMC

Reference Numbers : 1

Current used voice entity : 13

Voice entities are offered :

13 11

#

End

Table 11 Output description

Field Description

ViIfIndex Index of the voice interface from which the call is originated

Module ID ID of a voice module that the call passes through

Call direction Call direction of the call

13

Field Description

Reference Numbers Number of times of referencing the call information table of a call

entity 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.







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!

#

14

*************** CCB[1] ***************

GblCallID : 0x10000

CalledAddr : 2961

CalledAddrSubst : 2961

CallerAddr :

CallerAddrSubst :

CallInfoTabIndex : 0

Call Leg Number : 2

Active Service : 0

INCOMING CALLLEG NUMBER : 1

INCOMING LEG[0]

{

Spl Protocol : LGS

LocalRef : 0x0002

IfIndex : 2884067

IpAddress : 0.0.0.0

IpPort : 0

LegState : IN_STATE_ACTIVE

ConnectState : CONN_STATE_ACTIVE

}

OUTGOING CALLLEG NUMBER : 1

OUTGOING LEG[0]

{

Spl Protocol : LGS

LocalRef : 0x0003

IfIndex : 2884064

IpAddress : 0.0.0.0

IpPort : 0

LegState : OUT_STATE_ACTIVE

ConnectState : CONN_STATE_ACTIVE

}

#

End.

# 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 : 0

Send SETUP_ACK message : 0

Send ALERTING message : 0

Send CONNECT message : 0

Send RELEASE message : 0

Send RELEASE_COMP message : 0

Send INFORMATION message : 0

Send SWITCH_CODEC message : 0

Send FAXVOC_SWTH message : 0

Send FAXVOC_SWTHACK message : 0

15

Receive SETUP message : 0

Receive SETUP_ACK message : 0

Receive ALERTING message : 0

Receive CONNECT message : 0

Receive RELEASE message : 0

Receive RELEASE_COMP message : 0

Receive INFORMATION message : 0

Receive SWITCH_CODEC message : 0

Receive FAXVOC_SWTH message : 0

Receive FAXVOC_SWTHACK message: 0

}


Field Description

GblCallID Indicates the global ID of the call.

CalledAddr Indicates the called number of the call.

CalledAddrSubst Indicates the called number after substitution.

CallerAddr Indicates the caller number of the call.

CallerAddrSubst Indicates the caller number after substitution.

CallInfoTabIndex Indicates the call information index of the call.

Call Leg Number Indicates the number of call legs of the call.

Active Service Indicates the number of services involved in the call.

Spl Protocol Indicates the type of protocol used in the call leg.

LocalRef Indicates the local call identifier of the call leg.

IfIndex Indicates the voice interface index connected to the call leg.

IpAddress Indicates the IP address connected to the call leg.

IpPort Indicates the port number connected to the call leg.

LegState Indicates the state of the call leg.

ConnectState Indicates the state of connection of the call.

SETUP message Statistics of SETUP messages sent to or from the LGS module

SETUP_ACK message Statistics of SETUP_ACK messages sent to or from the LGS module

ALERTING message Statistics of ALERTING messages sent to or from the LGS module

CONNECT message Statistics of CONNECT messages sent to or from the LGS module

RELEASE message Statistics of RELEASE messages sent to or from the LGS module

RELEASE_COMP message Statistics of RELEASE_COMP messages sent to or from the LGS module

INFORMATION message Statistics of INFORMATION messages sent to or from the LGS module

SWITCH_CODEC message Statistics of SWITCH_CODEC messages sent to or from the LGS module

FAXVOC_SWTH message Statistics of FAXVOC_SWTH messages sent to or from the LGS module

FAXVOC_SWTHACK message Statistics of FAXVOC_SWTHACK messages sent to or from the LGS module

16

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







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 20(system: 20)


default entity payload-size g726r16 30(system: 30)





17

default entity modem compatible-param 100(system: 100)

default entity modem protocol pcm disable


Field Description

fax ecm ECM mode is used for Fax.

fax protocol t38 Fax protocol for intercommunication

fax redundancy t38 hb-redundancy Number of high-speed redundant packets, available for standard

T.38 or T.38

fax redundancy t38 lb-redundancy Number of low-speed redundant packets, available for standard T.38

or T.38

fax level Gateway carrier transmitting energy level

fax local-train threshold Fax local training threshold percentage

fax baudrate Highest Fax rate

fax nsf-on Fax capacity negotiation mode

fax train-mode Fax training mode

fax cng-switch CNG fax switch

compression 1st-level Voice coding mode of the first preference

compression 2nd-level Voice coding mode of the second preference

compression 3rd-level Voice coding mode of the third preference

compression 4th-level Voice coding mode of the fourth preference

vad-on Voice entity VAD

payload-size g711 Voice entity packet assembly interval (G.711)


payload-size g726r16 Voice entity packet assembly interval (G.723 r16)





modem compatible-param Value of the payload type field for the NTE-compatible switching

mode.

modem protocol pcm 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

18

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.







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


Field Description

Current configuration of entities Configured voice entities

entity 66 pots POTS voice entity numbered 66

match-template Template for number matching

line Voice subscriber line bound to the voice entity

19

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 }


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.







Examples

# Display statistics about H.225 messages of the IPP module.

<Sysname> display voice ipp statistic h225

Statistics about H225 :

{

Send_Setup : 0

Send_CallProceeding : 0

Send_Alerting : 0

Send_Connect : 0

Send_ReleaseComplete : 0

Send_FacilityIndUserInput : 0

Send_FacilityTCSRequest : 0

Send_FacilityTCSAck : 0

Send_FacilityTCSReject : 0

Send_FacilityOLCRequest : 0

Send_FacilityOLCAck : 0

20

Send_FacilityOLCReject : 0

Send_FacilityMSDRequest : 0

Send_FacilityMSDAck : 0

Send_FacilityMSDReject : 0

Send_FacilityCLCRequest : 0

Send_FacilityCLCAck : 0

Send_FacilityStartH245 : 0

Send_Error : 0

Recv_Setup : 0

Recv_CallProceeding : 0

Recv_Alerting : 0

Recv_Connect : 0

Recv_ReleaseComplete : 0

Recv_Progress : 0

Recv_FacilityTCSRequest : 0

Recv_FacilityTCSAck : 0

Recv_FacilityTCSReject : 0

Recv_FacilityOLCRequest : 0

Recv_FacilityOLCAck : 0

Recv_FacilityOLCReject : 0

Recv_FacilityMSDRequest : 0

Recv_FacilityMSDAck : 0

Recv_FacilityMSDReject : 0

Recv_FacilityCLCRequest : 0

Recv_FacilityCLCAck : 0

Recv_Unknown : 0

}


Field Description

Setup Statistics of Setup messages

CallProceeding Statistics of CallProceeding messages

Alerting Statistics of Alerting messages

Connect Statistics of Connect messages

ReleaseComplete Statistics of ReleaseComplete messages

FacilityIndUserInput Statistics of UserInput messages

FacilityTCSRequest Statistics of TCS Request messages

FacilityTCSAck Statistics of TCS Acknowledgement messages

FacilityTCSReject Statistics of TCS Reject messages

FacilityOLCRequest Statistics of OLC Request messages

FacilityOLCAck Statistics of OLC Acknowledgement messages

FacilityOLCReject Statistics of OLC Reject messages

FacilityMSDRequest Statistics of MSD Request messages

21

Field Description

FacilityMSDAck Statistics of MSD Acknowledgement messages

FacilityMSDReject Statistics of MSD Reject messages

FacilityCLCRequest Statistics of CLC Request messages

FacilityCLCAck Statistics of CLC Acknowledgement messages

FacilityStartH245 Statistics of H.245 Start messages

Error Statistics of Error messages

Unknown 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 }


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 timer‘s statistic information of the IVA module.

vim: Displays all the interaction statistic information between IVA module and VIM.







Examples

# Display the call statistics between IVA module and other modules.

<Sysname> display voice iva statistic call

22

Statistics about IVA calls :

{

IVA_ISDN_ACTIVE_CALL : 0

IVA_ISDN_ACTIVE_CALL_SUCCEEDED : 0

IVA_ISDN_ACTIVE_CALL_FAILED : 0

IVA_ISDN_PASSIVE_CALL : 0

IVA_ISDN_PASSIVE_CALL_SUCCEEDED : 0

IVA_ISDN_PASSIVE_CALL_FAILED : 0

}


Field Description

IVA_ISDN_ACTIVE_CALL Statistics of calls generated when IVA serves as the caller

IVA_ISDN_ACTIVE_CALL_SUCCEEDED Statistics of successful calls when IVA serves as the caller

IVA_ISDN_ACTIVE_CALL_FAILED Statistics of failed calls when IVA serves as the caller

IVA_ISDN_PASSIVE_CALL Statistics of calls generated when IVA serves as the called

IVA_ISDN_PASSIVE_CALL_SUCCEEDED Statistics of successful calls when IVA serves as the called

IVA_ISDN_PASSIVE_CALL_FAILED 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.



23





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

24

IfIndex:0x2c00c0


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



End


Field Description

SetupTime The length of the time from the system starts up to the start time

of the call, in milliseconds.

Index Identification number, which defaults to 1. For the records with

the same Setup Time, their index values increase by degrees.

PhoneSubNumber Sub-number of a phone. Not supported.

EntityIndex Entity identification number. If the entity does not exist, the entity

index is null.

IfIndex Index number of the interface of the voice subscriber line

corresponding to the entity.

ConnectTime Accumulated connect time to the peer since the system started

up, in milliseconds.

CallState

Call state:

Unknown: The call state is unknown.

Connecting: A connection attempt (outgoing call) is being

made.

Connected: A connection attempt (incoming call) is being

made.

Active: The call is active.

25

Field Description

CallOrigin Role in a call, originate or answer.

ChargedUnits Number of charged units for a connection; not supported.

CallInfoType Information type for this call, Speech or Fax.

ByteReceived Number of the received bytes. The maximum value is

4,294,967,295.

ByteTransmited Number of the transmitted bytes. The maximum value is

4,294,967,295.

PacketReceived Number of the received packets. The maximum value is

4,294,967,295.

PacketTransmited Number of the transmitted packets. The maximum value is

4,294,967,295.

ConnectionId Connection ID, which is used to identify a call.

CallId Identification number of the calling side.

RemoteSignallingIPAddress IP address of the remote signaling.

RemoteSignallingPort Port number of the remote signaling.

RemoteMediaIPAddr IP address of the remote media.

RemoteMediaPort Port number of the remote media.

SessionProtocol Session protocol type. Only the SIPv2 protocol is supported.

CallingNumber Calling number before the substitution.

CalledNumber Called number before the substitution.

SubstCallingNumber Substituted calling number.

SubstCalledNumber Substituted called number.

TxDuration Open duration of a call link, the open duration of the media

channel, in milliseconds.

VoiceTxDuration

Transmission duration of voice data, in milliseconds.

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.

FaxTxDuration Duration of fax transmission, in milliseconds.

For multiple times of fax, the values are added.

ImgPages Number of pages faxed.

For multiple times of fax, the value is added.

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.

26

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.







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


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

27

ImgPages:0

CodecType:G729r8

CallingNumber:6001

CalledNumber:7001



Call-History Info:

Index:2

SetupTime:155448 ms

PhoneNumber:6001

EntityIndex:6000

IfIndex:0x0


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



End


Field Description

SetupTime Length of the time from the system starts up to the start time of the

call, in milliseconds.

EntityIndex Entity identification number. If the entity does not exist, the entity

index is null.

IfIndex Index number of the interface of the voice subscriber line

corresponding to the entity.

28

Field Description


up, in milliseconds.

TerminateTime The length of the time from when the system starts up to when

the terminate time of the call, in milliseconds.

CallOrigin Role in a call, originate or answer.

ChargedUnits Number of charged units for a connection; not supported.

CallInfoType Information type for this call, Speech or Fax.

CallId Identification number of the calling side.

RemoteSignallingIPAddress The remote signaling IP address.

RemoteSignallingPort The remote signaling port number.

RemoteMediaIPAddr The remote media IP address.

RemoteMediaPort The remote media port number.

SessionProtocol Session protocol type. Only SIPv2 is supported.

CallingNumber Calling number before the substitution.

CalledNumber Called number before the substitution.

SubstCallingNumber Substituted calling number.

SubstCalledNumber Substituted called number.

ConnectionId Connection ID, which is used to identify a call.

TxDuration Open duration of a call link, the open duration of the media

channel, in milliseconds.

VoiceTxDuration

Transmission duration of voice data, in milliseconds.

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.

FaxTxDuration Duration of fax transmission, in milliseconds.


ImgPages Number of pages faxed.


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.

29

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.







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

30


Field Description

Index Entity index

Type Entity type, which can be POTS, VoIP, or Other.

Match-Template Number template


up, in milliseconds

LastSetupTime 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-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.

31

Use undo dscp media to restore the default DSCP.

By default, the DSCP value is ef (101110).

Syntax

dscp media dscp-value

undo dscp media

View


Default level

2: System level

Parameters

dscp-value: DSCP value in the range of 0 to 63 or the keyword af11, af12, af13, af21, af22, af23, af31, af32,

af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, and ef.

Examples

# Set the DSCP value in the ToS field of the IP packets that carry the RTP stream of VoIP voice entity to af41.





[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


Default level

2: System level

32

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.





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-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.

33

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


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 Meaning

0-9 Numbers from 0 to 9. Each means a digit.

# and * Each means a valid digit.

. A wildcard. It can match any digit of a valid number. For example, 555. . . . matches any

string that begins with 555 and with four additional characters.

! The character or characters right in front of it does not appear or appears once. For

example, 56!1234 can match 51234 and 561234.

34

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.

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

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.

Examples

# Specify 5557922 as a telephone number of voice entity 10.





[Sysname-voice-dial-entity10] match-template 5557922

# Configure a match template for VoIP voice entity 10.






35

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-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 { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } time-length

undo payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 }

36

View


Default level

2: System level

Parameters

g711: Packetization period in milliseconds for g711alaw or g711ulaw codec, an integral multiple of 10 in the

range of 10 to 30, with a default of 20.

g723: Packetization period in milliseconds for g723r53 or g723r63 codec, an integral multiple of 30 in the


g726r16: Packetization period in milliseconds for g726r16 codec, an integral multiple of 10 in the range of

10 to 110, with a default of 30.







g729: Packetization period in milliseconds for g729r8 or g729a codec, an integral multiple of 10 in the


time-length: DSP packetization period for a codec.

Examples

# Set the voice packetization period of the DSP for g711 codec to 30 milliseconds.





[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.

37

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-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.

38

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.

39

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-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

40

Parameters

None

Examples

# Enable the local end to play ringback tone.





[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


Default level

2: System level

Parameters

None

Examples

# Change the management status of voice entity 4 to DOWN.





[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.

41

The G.711 and G.726 codecs do not support VAD.

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.


g729a: Specifies the g729a codec.


Examples

# Enable VAD on POTS voice entity 10.





[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

42

Parameters

None

Examples

# Enter voice view and enable voice services.



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-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, 101110.

43

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 af11, af12, af13, af21, af22, af23, af31, af32,

af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, or ef.

Table 21 DSCP values

Keyword DSCP value in binary DSCP value in decimal

af11 001010 10

af12 001100 12

af13 001110 14

af21 010010 18

af22 010100 20

af23 010110 22

af31 011010 26

af32 011100 28

af33 011110 30

af41 100010 34

af42 100100 36

af43 100110 38

cs1 001000 8

cs2 010000 16

cs3 011000 24

cs4 100000 32

cs5 101000 40

cs6 110000 48

cs7 111000 56

default 101110 46

44

Keyword DSCP value in binary DSCP value in decimal

ef 101110 46

Examples

# Set the DSCP value in the ToS field in the IP packets that carry voice signaling to af41.



[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-voice] vqa dsp-monitor buffer-time 300

45

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-voice] area north-america

46

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] 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.

47

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


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-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.



48

[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


Default level

2: System level

Parameters

None

Examples

# Enable CID on voice 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 number‘s 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

49

View


Default level

2: System level

Parameters

None

Examples

# Enable CID on voice 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


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-subscriber-line1/0] cid ring 0

50

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-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

51

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-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-subscriber-line1/0] undo cng-on

52

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 Country name (including customization)

AR Argentina

AU Australia

AT Austria

BE Belgium

BR Brazil

BG Bulgaria

CA Canada

CL Chile

CN China

CS Customizes the call progress tones

HR Croatia

CU Cuba

53


CY Cyprus

CZ Czech Republic

DK Denmark

EG Egypt

FI Finland

FR France

DE Germany

GH Ghana

GR Greece

HK Hong Kong China

HU Hungary

IS Iceland

IN India

ID Indonesia

IR Iran

IE Ireland

IEU Ireland (UK style)

IL Israel

IT Italy

JP Japan

JO Jordan

KE Kenya

KR Korea Republic

LB Lebanon

LU Luxembourg

MO Macau

MY Malaysia

MX Mexico

NP Nepal

NL Netherlands

NZ New Zealand

NG Nigeria

NO Norway

PK Pakistan

PA Panama

54


PH Philippines

PL Poland

PT Portugal

RU Russian Federation

SA Saudi Arabia

SG Singapore

SK Slovakia

SI Slovenia

ZA South Africa

ES Spain

SE Sweden

CH Switzerland

TH Thailand

TR Turkey

GB United Kingdom

US United States

UY Uruguay

ZW Zimbabwe

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

55

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-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-subscriber-line5/0] default

This command will restore the default settings. Continue? [Y/N]:y

56

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-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

57

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-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.


Syntax

delay rising milliseconds

undo delay rising

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-subscriber-line5/0] em-signal delay

[Sysname-subscriber-line5/0] delay rising 700

58

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.


Syntax

delay send-dtmf milliseconds

undo delay send-dtmf

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-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.


Syntax

delay send-wink milliseconds

undo delay send-wink

View


Default level

2: System level

59

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-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.


Syntax

delay wink-hold milliseconds

undo delay wink-hold

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-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.

60

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.


Syntax

delay wink-rising milliseconds

undo delay wink-rising

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-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.



61

[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



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-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

62

Parameters

None

Examples

# Enable the sending of pulse signals at hangup on the FXS voice 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.







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 ----- subscriber-line5/0

Type = Analog E&M Immediate-Start

Status = UP

Call Status = BUSYTONE

Description = subscriber-line5/0 Interface

Private Line = None

Cng = Enable

63

Echo Canceller = Enable

Echo Canceller Tail-Length = 32

Nlp On = Enable

Receive Gain = 0.0

Transmit Gain = 0.0

DTMF Threshold Analogue :

Index 0 = 1400

Index 1 = 458

Index 2 = -9

Index 3 = -9

Index 4 = -9

Index 5 = -9

Index 6 = -3

Index 7 = -12

Index 8 = -12

Index 9 = 30

Index 10 = 300

Index 11 = 3200

Index 12 = 375

Timer Dial-Interval = 10

Timer Wait-Digit = 5

Timer Ring-Back = 60

Delay Send-dtmf = 300

E&M Physical Wire = 4-Wire

E&M Type = V

Slic-Gain = 0.8 db

Physical Information :

Card Type = E&M

Physical State = 1

Logical State = 1

Voice State = Uninstall

ResetCount = 0

InPkts = 0

OutPkts = 0

InBytes = 0

OutBytes = 0

LastRcvPacketLen = 0

LastSndPacketLen = 0

CmdInBuff = 0

CmdInTotalBuff = 0

DataInBuff = 0

DataInTotalBuff = 0

AbortCmdCount = 0

AbortPktsCount = 0

G723R53ToR63Packet = 0

G723R63ToR53Packet = 0

ClearDspBuffCount = 0

64


Field Description

Type Type of voice subscriber line

Status Status of voice subscriber line

Call Status Call status of voice subscriber line

Description Description of voice subscriber line

Private-line Private line dial number of voice subscriber line

CNG Comfortable noise configuration on voice subscriber line

EchoCancel Echo duration configuration on voice subscriber line

Nlp-on Non-linear process of echo cancel on voice subscriber line

Receive gain Receive gain configuration on voice subscriber line

Transmit gain Transmit gain configuration on voice subscriber line

DTMF Threshold Analogue DTMF threshold configuration of analog voice subscriber line

Timer Dial-Interval Dial interval of voice subscriber line

Timer Wait-Digit Period of timeout waiting for a number on voice subscriber line

Timer Ring-Back Period of timeout when ringing back on voice subscriber line

Delay Send-dtmf Pre-dial delay of voice subscriber line

E&M Physical Wire Cable type of analog E&M voice interface

E&M Type Circuit type of analog E&M voice interface

Slic-Gain SLIC gain configuration of analog E&M voice interface

Physical Information Physical statistics information

Card Type Type of the voice interface card

Physical State Physical state of the voice interface

Logical State Logical state of the voice interface

Voice State Call state on the voice interface

ResetCount Indicates how many times the voice interface card is reset

InPkts Number of received packets on the voice interface

OutPkts Number of sent packets on the voice interface

InBytes Bytes of received packets on the voice interface

OutBytes Bytes of sent packets on the voice interface

LastRcvPacketLen Length of the last received packet on the voice interface

LastSndPacketLen Length of the last sent packet on the voice interface

CmdInBuff Number of commands in the command buffer of the voice interface

CmdInTotalBuff Total number of commands in the command buffers of the voice interface card

AbortCmdCount Number of command packets discarded on the voice interface

AbortPktsCount Number of packets discarded on the voice interface

65

Field Description

G723R53ToR63Packet Number of G723R53 packets converted to G723R63 packets on the voice

interface

G723R63ToR53Packet Number of G723R63 packets converted to G723R53 packets on the voice

interface

ClearDspBuffCount 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-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:

66

Command A-MSR900 A-MSR20-1X A-MSR20 A-MSR30 A-MSR50

dtmf sensitivity-level

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-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

67

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-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.

68

Table 24 Meaning of the index numbers

Index Meaning Value range Remarks

0

Lower limit of (ROWMAX + COLMAX). The input

signal which is otherwise regarded too weak is

recognized as a DTMF digit when ROWMAX +

COLMAX) > 0.

1 to 4,999, with a

default of 1,400

The larger the value is,

the higher the detection

reliability is. However,

the sensitivity decreases.

1

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.

1 to 4,999, with a

default of 458

The smaller the value is,




2

Lower limit of COLMAX/ROWMAX, where

ROWMAX < COLMAX. An input signal is

recognized as a DTMF digit only when 10 x

(COLMAX/ROWMAX) > 2.

–18 to –3 dB, with a

default of –9 dB





3

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.


default of –9 dB





4

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.


default of –9 dB





5

Upper limit of the ratio of the second largest

energy level from the column frequency group to

COLMAX. The ratio must be lower than this limit

for the input signal to be recognized as a DTMF

digit.


default of –9 dB





6

Upper limit of ROW2nd/ROWMAX. An input

signal is recognized as a DTMF digit only when

ROW2nd/ROWMAX < 6.


default of –3 dB





7

Upper limit of COL2nd/COLMAX. The ratio must

be lower than this limit for the input signal to be

recognized as a DTMF digit.


default of –12 dB





8

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.


default of –12 dB





9

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.

30 to 150

milliseconds, with a

default of 30

milliseconds





69

Index Meaning Value range Remarks

10

Frequency of the first extra frequency point

specified for detection.

In addition, it must be a frequency 100 Hz

greater than or less than the row and column

frequency groups.

300 to 3,400 Hz,

with a default of 300

Hz

—

11

Frequency of the second extra frequency point

specified for detection.

In addition, it must be a frequency 100 Hz

greater than or less than the row and column

frequency groups.

300 to 3,400 Hz,

with a default of

3,200 Hz

—

12

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.

0 to 700, with a

default of 375





Examples

# Set the DTMF threshold 9 for voice 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



Default level

2: System level

Parameters

enable: Enables the echo cancellation (EC) function.

70

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-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 511. The

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-voice] echo-canceller parameter convergence-rate 50

71

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-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


Default level

2: System level

72

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-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


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-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.

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Syntax

hookoff-mode { delay | immediate }

undo hookoff-mode

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-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


74

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-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


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-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.

75

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.









Examples

# Set the current electric impedance to r600 on voice 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.


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nlp-on

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

NOTE:

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

Syntax

nlp-on

undo nlp-on

View


Default level

2: System level

Parameters

None

Examples

# Disable the EC nonlinear processing function on voice interface.



[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.

77

Syntax

open-trunk { caller monitor interval | called }

undo open-trunk

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-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.



78

[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



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-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

79

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

80

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.



ring-detect debounce

The following voice modules support this command:

SIC-2FXO

SIC-1FXO

MIM-4FXO

MIM-2FXO

FIC-4FXO

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


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] 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.



ring-detect frequency The following voice modules support this command:

SIC-2FXS1FXO

Syntax

ring-detect frequency value

undo ring-detect frequency

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] 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


82

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-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



Default level

2: System level

Parameters

None

Examples

# Shut down voice subscriber line 1/0.



[Sysname-subscriber-line1/0] shutdown

83

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-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


Default level

2: System level

84

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-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-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

85

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-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-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.

86

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-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-subscriber-line1/0] timer hookflash-detect 100-200

87

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


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-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

88

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-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


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-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.

89

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



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-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


Default level

2: System level

Parameters

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

90

Examples

# Configure subscriber line 5/0 analog E&M subscriber line type as type 3.



[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.

91

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-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-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.

92

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-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.



93

[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] 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 }

94

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-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

95

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] 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.


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.

96

Examples

# Start requesting calling number or caller identifier after receiving three digits.





[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.


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-cas1/0:0] undo answer enable

callmode Description

Use callmode to configure the connection mode for an R2 call.

97

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-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.


98




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.


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-cas1/0:0] clear-forward-ack enable

99

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.







Examples

# Display the configuration of voice subscriber line 5/0:0.

<Sysname> display voice subscriber-line 5/0:0

Current information ----- subscriber-line5/0:0

Type = R2

Status = PhysicalDown

Call Status :

TS 1 = IDLE

TS 2 = IDLE

TS 3 = IDLE

TS 4 = IDLE

TS 5 = IDLE

TS 6 = IDLE

TS 7 = IDLE

TS 8 = IDLE

TS 9 = IDLE

TS 10 = IDLE

100

TS 11 = IDLE

TS 12 = IDLE

TS 13 = IDLE

TS 14 = IDLE

TS 15 = IDLE

TS 17 = IDLE

TS 18 = IDLE

Description = subscriber-line5/0:0 Interface

Private Line = None

Cng = Enable

Echo Canceller = Enable

Echo Canceller Tail-Length = 32

Nlp On = Enable

Receive Gain = 0.0

Transmit Gain = 0.0

DTMF Threshold Digital = Insensitivty

PCM Type = A-Law


Field Description

Current information Information about the current voice subscriber line

Type Signaling type on the voice subscriber line

Status Status of the voice subscriber line

Call Status Status of the voice protocol call

Description Information about the voice subscriber line

Private Line Private line dialup mode of the voice subscriber line

Cng Comfort noise setting on the voice subscriber line

The subscriber line's description The description of the subscriber line

Echo Canceller Echo cancellation setting on the voice subscriber line

Echo Canceller Tail-Length Echo interval setting on the voice subscriber line

Nlp on Setting of nonlinear processing (NLP) in the echo canceller

on the voice subscriber line

Receive Gain Input gain of the voice subscriber line

Transmit Gain Output gain of the voice subscriber line

DTMF Threshold Digital DTMF parameters of the digital voice subscriber line

PCM Type 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.

101

Use undo dl-bits to restore the defaults.


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 1111.

Table 26 Default values of signals in R2 digital line signaling

Signal Default rx-bits ABCD Default tx-bits ABCD

Answer 0101 0101

Blocking 1101 1101

Clear-back 1101 1101

Clear-forward 1001 1001

Idle 1001 1001

Seize 0001 0001

Seizure-ack 1101 1101

Release-guard 1001 1001

102

Examples

# Set the ABCD bit pattern for received R2 idle signal to 1101, and to 1011 for transmitted R2 idle signal.





[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.


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-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.

103

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] 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] interface bri 2/0

104

[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.


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-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

105

Parameters

None

Examples

# Enable R2 metering signal processing.





[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.


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-cas1/0:0] group-b enable

106

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


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-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.


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

107

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-cas1/0:0] mode hongkong default-standard

108

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 µ-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 µ-law.

Syntax

pcm { a-law | µ-law }

undo pcm

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.

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

Examples

# Adopt µ-law companding for signal quantization.



[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

109

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-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


Default level

2: System level

110

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-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-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.


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

111

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-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.


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.

112

Syntax

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 } 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.

113

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 subscriber‘s line abnormal signal value, in the range of 1 to 16.

subscriber-busy value: Specifies the subscriber‘s line busy signal value, in the range of 1 to 16.

subscriber-charge value: Specifies the charge value when the subscriber‘s line is idle, in the range of 1 to 16.

subscriber-idle value: Specifies the subscriber‘s line idle value, in the range of 1 to 16. It configures the R2

KB signal used for describing the called subscriber‘s 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-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.


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

11 in some other countries.

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

114

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 11.





[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.


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.




115


[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.


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-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.

116

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-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.


Related commands: timer ring.

Syntax

sendring ringbusy enable

undo sendring ringbusy enable

117

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-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 1101, 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.

118

Examples

# Set the ABCD bit pattern to 1011 for the transmit seized signal on digital E&M subscriber line 1/0:0.



[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.


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 11 to 16.

Examples

# Assign the pound sign (#) the register signal code 11.





[Sysname-cas1/0:0] special-character # 11

119

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-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:

120

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


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-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.


Syntax

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

121

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-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.

122

By default, the delay is 50 milliseconds.


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-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

123

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] 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.


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-cas1/0:0] timer register-pulse persistence 300

124

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.


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-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.


Related commands: sendring.

Syntax

timer ring { ringback | ringbusy } time

undo timer ring { ringback | ringbusy }

125

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-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


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.

126

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.




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.


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.

127

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-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.




128


[Sysname-cas1/0:5] ts reset timeslots 1-15

[Sysname-cas1/0:5] ts query timeslots 1-31

129

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-voice-dial-entity1] caller-group permit 1

caller-permit Description

Use caller-permit to configure a calling number permitted to originate calls.

130

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 Meaning

0-9 Digits 0 through 9.

# and * 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.

131

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-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

132

Parameters

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

Examples

# Identify subscriber group 10 as international.




[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 Meaning

0-9 Digits 0 through 9.

, One comma represents a pause of 500 milliseconds and it can be positioned

anywhere in a number.

# or * Indicates a valid digit each.

133

Examples

# Specify 0 as a prefix.





[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 }


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.







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..

#

134

END

Current configuration of subscriber group 2

#

Description : <NULL>

Referenced by entities:



Include match templates:

Match-template: 1200..

#

END


Field Description

Current configuration of the appointed

subscriber group Configuration information of a specified subscriber group

Description Description of a subscriber group

Referenced by entities Information of voice entities that a subscriber group is bound

to

Type Type of the voice entity that a subscriber group is bound to

Tag Tag of the voice entity that a subscriber group is bound to

Match-template 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.





135



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 : 4

First-rule : INDEX_INVALID

Dot-match : left-right

rule 1

Input-format : ^011408

Output-format : 1408

#

End

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.

136

Examples

# Set the dot match rule of number substitution rule list 20 to right-left.




[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


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-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.

137

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:



$: 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 Meaning

0-9 Digits from 0 through 9.

# and * 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.

138

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-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


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.

139

Examples

# Set the maximum number of call connections in maximum-call-connection set 1 to 5.




[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-voice-dial] max-call 1 5


[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.

140

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


Default level

2: System level

Parameters

longest: Matches the longest number.

shortest: Matches the shortest number.

Examples

# Configure the longest-number match mode.




[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


Default level

2: System level

Parameters

None

141

Examples

# Configure a number to first match against a voice entity match template.




[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


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-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.

142

Related commands: select-rule.

Syntax

priority priority-order

undo priority

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-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



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.


143

[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


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.



$: 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 Meaning

0-9 Digit 0 through 9.

# and * 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.

144

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.

2. 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.

3. 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


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


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.

145

Table 32 Input number type

Number type Description

abbreviated Abbreviated number

any Any number

international International number

national National number, but not a local network

network Specific service network number

reserved Reserved number

subscriber Local network number

unknown Number of an unknown type

output-number-type: Type of an output number involved in number substitution. For the values, see Table 33.

Table 33 Output number type

Number type Description

abbreviated Abbreviated number

international International number

national National number, but not a local network number

network Specific service network number

reserved Reserved number

subscriber Local network number

unknown 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 Description

any Any numbering plan

data Data numbering plan

isdn ISDN telephone numbering plan

national National numbering plan

private Private numbering plan

reserved Reserved numbering plan

telex Telex numbering plan

unknown Unknown numbering plan

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

see Table 35.

146

Table 35 Output numbering plan

Numbering plan Description

data Data numbering plan

isdn ISDN telephone numbering plan

national National numbering plan

private Private numbering plan

reserved Reserved numbering plan

telex Telex numbering plan

unknown Unknown numbering plan

Examples

# Configure number substitution rules for number substitution rule list 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


Input number: 92

Output number: 2



Input number: 93

Output number: 3



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

147

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


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 Meaning Description

1 Exact match

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.

2 Priority

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.

148

Integer Meaning Description

3 Random selection The system selects at random a voice entity from a set of

qualified voice entities.

4 Longest idle time 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-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


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-voice-dial] select-rule search-stop 5

149

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


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 Meaning

1 POTS voice entity

2 VoIP voice entity

3 VoFR voice entity

4 IVR voice entity

Examples

# Configure the system to select voice entities in the order of VoIP->POTS->VoFR->IVR.



150


[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


Default level

2: System level

Parameters

None

Examples

# Disable the voice entity search function for voice entity 10.





[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 }

151

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-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


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.

152





[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-voice] subscriber-line1/0

[Sysname-voice-line1/0] substitute called 6

153

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


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-voice-dial] substitute incoming-call called 5

# Apply number substitution rule lists 5, 6, and 8 to called numbers of outgoing calls.


154



[Sysname-voice-dial] substitute outgoing-call called 5



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


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-voice-dial] terminator #

155

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.






# Configure the domain name of the called entity as cc.news.com for voice entity 10.



156



[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-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.

157

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-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







158

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 44 0 0 0 44

OutMsgSucc 33 0 0 0 33

OutMsgFail 0 0 0 0 0

TXN Message Inv_Cli NonInv_Cli Inv_Srv NonInv_Srv

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

Create Succ 10 12 0 1

Create Fail 0 0 0 0

Terminal Abnom 0 0 0 0

Request Message Inv Ack Bye Can Opt Reg Inf Prk Upd

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

In: 0 0 1 0 0 0 0 0 0

Out: 10 10 4 3 0 5 0 0 0

Response Message 1xx 2xx 3xx 4xx 5xx 6xx

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

In: 21 13 0 9 0 0

Out: 0 1 0 0 0 0

Error Statistics:

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

callCb creation failures: 0

call-leg creation failures: 0

transaction creation failures: 0

callCb locate failures: 0

call-leg locate failures: 0

transaction locate failures: 0

user not registered: 0

user not available: 0

request with missing headers: 0

response-no To tag in response: 0

response - invalid via: 0

messages without headers rcvd: 0

SDP decode failures: 0

registration timeouts: 0

retransmitted requests received: 0

transaction timeouts: 0

159


Field Description

TPT Message

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)

TXN Message

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)

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)

Request Message

Statistics of all SIP request messages, including Inv (INVITE), ACK , BUE,

Can (CANCEL), Opt (OPTIONS), Reg (GEGISTER), Inf (Information), Prk

(PRACK), Upd (UPDATE)


In (received)

Out (sent)

Response Message

Statistics of all SIP response messages, including 1XX, 2XX, 3XX, 4XX

(Cancel), 5XX, and 6XX


In (received)

Out (sent)

callCb creation failures Statistics of call control block creation failures in SIP

call-leg creation failures Statistics of call leg creation failures in SIP

transaction creation failures Statistics of transaction creation failures in SIP

callCb locate failures Statistics of call control block location failures in SIP

call-leg locate failures Statistics of call leg location failures in SIP

transaction locate failures Statistics of transaction location failures in SIP

user not registered Statistics of user not registered message in SIP

user not available Statistics of user not available message in SIP

request with missing headers Statistics of request messages with missing headers in SIP

response-no To tag in response Statistics of response messages without the To Tag field in SIP

response - invalid via Statistics of response messages with an invalid via field in SIP

messages without headers rcvd Statistics of received messages without headers in SIP

SDP decode failures Statistics of SDP decoding failures in SIP

registration timeouts Statistics of registration timeouts in SIP

retransmitted requests received Statistics of received retransmission requests in SIP

160

Field Description

transaction timeouts 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.







Examples

# Display the information about SIP connections over TCP.

<Sysname> display voice sip connection tcp

Conn-Id Local-IP Local-Port Remote-IP Remote-Port Conn-State

+------------------------------------------------------------------------------+

569 100.1.1.84 1593 100.1.1.100 5060 Established

570 100.1.1.84 1594 100.1.1.101 5060 Established

571 100.1.1.84 1595 100.1.1.81 5060 Established

572 192.168.0.82 1596 192.168.0.81 5060 Established

# Display the information about SIP connections over TLS.

<Sysname> display voice sip connection tls

Conn-Id Local-IP Local-Port Remote-IP Remote-Port Conn-State

+------------------------------------------------------------------------------+

73 192.168.0.202 1086 192.168.0.132 5061 Established

161

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.







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


Field Description

Current configuration of ENUM groups Configuration information of ENUM translation rule groups.

enum-group 1 ENUM translation rule group.

rule 1 preference 1 408…(8333) 5555\1

cc.news.com ENUM translation rule.

162

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.







Examples

# Display SIP DNS records.

<Sysname> display voice sip dns-record

No. Host IP

1 sip1.hp.com 100.1.1.163:5060

2 sip2.8056.com 100.1.1.16:5060


Field Description

No. Sequence number of the DNS record

Host Domain name

IP 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 ]

163

View

Any view

Default level

2: System level

Parameters







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 PSTN-Reason SIP-Status Default

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

1 1 400* 404

2 2 404 404

3 3 404 404

4 16 --- ---

5 17 486 486

6 18 408 408

7 19 480 480

8 20 480 480

9 21 403 403

10 22 410 410

11 23 410 410

12 25 500 500

13 26 404 404

14 27 502 502

15 28 484 484

16 29 501 501

17 31 480 480

18 34 503 503

19 38 503 503

20 41 503 503

21 42 503 503

22 47 503 503

23 55 403 403

24 57 403 403

164

25 58 503 503

26 63 500 500

27 65 488 488

28 70 488 488

29 79 501 501

30 87 403 403

31 88 503 503

32 102 504 504

33 111 500 500

34 127 500 500


Field Description

PSTN-Reason PSTN release cause code

SIP-Status

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 Default SIP status code corresponding to a PSTN release cause

code

# 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 SIP-Status PSTN-Reason Default

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

1 400 127* 41

2 401 21 21

3 402 21 21

4 403 21 21

5 404 1 1

6 405 63 63

7 406 79 79

8 407 21 21

9 408 102 102

10 410 22 22

11 413 127 127

12 414 127 127

13 415 79 79

14 416 127 127

15 420 127 127

16 421 127 127

17 423 127 127

18 480 18 18

19 481 41 41

165

20 482 25 25

21 483 25 25

22 484 28 28

23 485 1 1

24 486 17 17

25 487 127 127

26 488 127 127

27 500 41 41

28 501 79 79

29 502 38 38

30 503 41 41

31 504 102 102

32 505 127 127

33 513 127 127

34 600 17 17

35 603 21 21

36 604 1 1

37 606 58 58


Field Description

SIP-Status SIP status code

PSTN-Reason

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 Default PSTN release cause code corresponding to a SIP status

code

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

166

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-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







Examples

# Display all registration status information on the SIP UA.

<Sysname> display voice sip register-state

Number Entity Registrar Server Expires Status

+-----------------------------------------------------------------------+

105 105 100.1.1.1:5060 N/A login

2000 107 100.1.1.1:5060 200 online

3000 109 cc.news.com:1120 N/A login

167


Field Description

Number User number

Entity Entity number

Registrar Server Address of the registrar, in the format of IP address + port number or domain

name + port number

Expires Aging time for a user number in seconds

Status

State in which a number stays, including:

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-voice] sip

[Sysname-voice-sip] undo early-media enable

168

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


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-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.

169

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-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-voice] sip

[Sysname-voice-sip] undo line-check

170

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-voice] sip

[Sysname-voice-sip] listen transport tls

171

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-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.


Syntax

outband sip

undo outband

View

POTS entity view, VoIP entity view

172

Default level

2: System level

Parameters

None

Examples

# Configure the out-of-band SIP DTMF transmission for VoIP entity 10.






[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 1120 of the outbound proxy server for the SIP UA.



[Sysname-voice] sip

[Sysname-voice-sip] outbound-proxy ipv4 169.54.5.10 port 1120

173

# Configure domain name abc.com and port number 1100 of the outbound proxy server for the SIP UA.



[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-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 ]

174

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 1120 for the proxy server.



[Sysname-voice] sip

[Sysname-voice-sip] proxy ipv4 169.54.5.10 port 1120

# Specify the domain name abc.com and port number 1100 for the proxy server.



[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

PSTN release cause description

SIP status code SIP status description

1 Unallocated (unassigned)

number! 404 Not Found

2 No route to specified transit

network! 404 Not Found

3 No route to destination! 404 Not Found

16 Normal clearing! --- BYE or CANCEL

17 User busy! 486 Busy here

18 No user responding! 408 Request Timeout

175




19 No answer from user! 480 Temporarily unavailable

20 Subscriber absent! 480 Temporarily unavailable

21 Call rejected! 403 Forbidden

22 Number changed! 410 Gone

23 Redirection to new

destination! 410 Gone

25 Exchange routing error! 500 Server internal error

26 Non-selected user clearing! 404 Not Found

27 Destination out of order! 502 Bad Gateway

28 Invalid number format

(address incomplete)! 484 Address incomplete

29 Facility rejected! 501 Not implemented

31 Normal, unspecified! 480 Temporarily unavailable

34 No circuit/channel

available! 503 Service unavailable

38 Network out of order! 503 Service unavailable

41 Temporary failure! 503 Service unavailable

42 Switching equipment

congestion! 503 Service unavailable

47 Resource unavailable,

unspecified! 503 Service unavailable

55 Incoming class barred within

Closed User Group (CUG)! 403 Forbidden

57 Bearer capability not

authorized! 403 Forbidden


presently available! 503 Service unavailable

63 Service or option not

available, unspecified! 500 Server internal error


implemented! 488 Not Acceptable Here

70

Only restricted digital

information bearer

capability is available!

488 Not Acceptable Here

79 Service or option not

implemented, unspecified! 501 Not implemented

87 User not member of Closed

User Group (CUG)! 403 Forbidden

88 Incompatible destination! 503 Service unavailable

102 Recovery on timer expiry! 504 Gateway timeout

176




111 Protocol error, unspecified! 500 Server internal error

127 Interworking, unspecified! 500 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-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 SIP status description PSTN release cause code


400 Bad Request 41 Temporary failure!

401 Unauthorized 21 Call rejected!

402 Payment required 21 Call rejected!

403 Forbidden 21 Call rejected!

404 Not found 1 Unallocated (unassigned)

number!

177



405 Method not allowed 63 Service or option not available,

unspecified!

406 Not acceptable 79 Service or option not

implemented, unspecified!

407 Proxy authentication

required 21 Call rejected!

408 Request timeout 102 Recovery on timer expiry!

410 Gone 22 Number changed!

413 Request Entity too long 127 Interworking, unspecified!

414 Request-URI too long 127 Interworking, unspecified!

415 Unsupported media type 79 Service or option not


416 Unsupported URI Scheme 127 Interworking, unspecified!

420 Bad extension 127 Interworking, unspecified!

421 Extension Required 127 Interworking, unspecified!

423 Interval Too Brief 127 Interworking, unspecified!

480 Temporarily unavailable 18 No user responding!

481 Call/Transaction Does not

Exist 41 Temporary failure!

482 Loop Detected 25 Exchange routing error!

483 Too many hops 25 Exchange routing error!

484 Address incomplete 28 Invalid number format (address

incomplete)!

485 Ambiguous 1 Unallocated (unassigned)

number!

486 Busy here 17 User busy!

487 Request Terminated 127 Interworking, unspecified!

488 Not Acceptable here 127 Interworking, unspecified!

500 Server internal error 41 Temporary failure!

501 Not implemented 79 Service or option not


502 Bad gateway 38 Network out of order!

503 Service unavailable 41 Temporary failure!

504 Server time-out 102 Recovery on timer expiry!

505 Version Not Supported 127 Interworking, unspecified!

513 Message Too Large 127 Interworking, unspecified!

600 Busy everywhere 17 User busy!

178



603 Decline 21 Call rejected!

604 Does not exist anywhere 1 Unallocated (unassigned)

number!

606 Not acceptable 58 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-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.

179

off: Disables the SIP registrar.

Examples

# Enable the SIP registrar.



[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-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.

180

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 1120, the registration aging time 120 seconds, and

the TCP transport layer protocol for the main registrar.



[Sysname-voice] sip

[Sysname-voice-sip] registrar ipv4 169.54.5.10 port 1120 expires 120 tcp

181

# Specify the domain name cc.news.com, the port number 1100, and the registration aging time 120

seconds of the main registrar.



[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-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.

182

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

183

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-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.

184

Syntax

sip

View

Voice view

Default level

2: System level

Parameters

None

Examples

# Enter SIP client view.



[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

185

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-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-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.


186


[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-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

187

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-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-voice] sip

[Sysname-voice-sip] source-bind signal ipv4 1.1.1.1

188

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-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

189

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-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-voice] sip

[Sysname-voice-sip] timer registration expires 600

timer registration divider Description

Use timer registration divider to set the registration percentage.

190

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-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.



191

[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-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

192

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-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


193

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-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.

194

sips: Specifies SIPS as the URL scheme for SIP calls.

Examples

# Specify SIPS as the global URL scheme for SIP calls.



[Sysname-voice] sip

[Sysname-voice-sip] url sips

# Configure SIPS as the URL scheme for the SIP calls on VoIP voice entity 1000.





[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.

195

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-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-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.

196

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-voice] sip

[Sysname-voice-sip] wildcard-register enable

197

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-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.

198

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-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-voice-server] call-route

199

[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-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.

200





[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-user1000] srs 1

# Apply a call rule set in sip server view.




[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.







Examples

# Display information of all users.

<Sysname> display voice sip-server register-user all

user number status address

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

201

1 404 online 192.168.0.98:5060

2 325 offline

3 380 online 192.168.0.57:5060


Field Description

User Tag of a user

Number Directory number of a user

Status

Registration status of a user, including

Offline

Online

Address 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







Examples

# Display the server resource information.

<Sysname> display voice sip-server resource-statistic

SIP Server state: Active

CbType Total Used Free

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

SLC_Conf 64 0 64

SLC_Call 128 0 128

SLC_Sub 64 0 64

202

SLC_Reg 64 0 64

SSA_Call 128 0 128

SSA_Sub 128 0 128


Field Description

SIP Server state

State of local SIP server:

Active

Inactive

CbType Type of the resource control module

Total Total number of resource control modules

Used Number of used resource control modules

Free Number of free resource control modules

SLC_Conf SLC control module

SLC_Call SLC call module

SLC_Sub SLC subscription module

SLC_Reg SLC regisration module

SSA_Call SSA call module

SSA_Sub 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

203

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-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–voice-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.

204

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-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.

205

Examples

# Configure the keepalive probe.




[Sysname–voice-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-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 }

206

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-voice-server-set–svc1] 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.

207

Examples

# Create a call rule.






[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-voice-server] server-bind ipv4 192.168.0.92

server enable Description

Use server enable to enable the local SIP server.

208

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-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.

209




[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-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.

210

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-voice-server] trusted-point ipv4 100.1.1.125

211

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.

212

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-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


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-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.

213

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-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-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

214

View

Account view

Default level

2: System level

Parameters

None

Examples

# Disable 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


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-voice-dial-entity1] bind sip-trunk account 1

215

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


Default level

2: System level

Parameters

None

Examples

# Enable codec transparent transmission.





[Sysname-voice-dial-entity1] codec transparent


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

216

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-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







Examples

# Display SIP trunk account information.

<Sysname> display voice sip-trunk account

ID User Group Server Exp Status

1 1000 1 202.10.22.188:5060 120 Online

2 2000 1 abc.com:5060 400 Online

3 3000 1 abc.com:5060 N/A Logout


Field Description

ID SIP trunk account index

User Host user name

Group SIP server group index

Server Address or domain name of the registrar

217

Field Description

Exp 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

Status

Registration status of the SIP trunk account:

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.







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 A’s 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

218

Hot swap mode: Disabled


Field Description

Index Index of the SIP server group: SIP-URI/SIPS URI; transport layer

protocol

Keepalive mode

Keep-alive mode of the SIP server group, including:

Disabled

REGISTER

OPTIONS

Hot swap mode

Real-time switching function status of the SIP server group,

including

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.




219

[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-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

220

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-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


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.

221

Examples

# Specify that calls with a source host name starting with Bil are permitted on VoIP voice entity 1.





[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


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-voice-dial-entity3] match destination host-prefix ali

222

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


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-voice-dial-entity3] match source address ipv4 100.1.1.*

223

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-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

224

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-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.




225

[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-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 }

226

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-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-voice-account-2]

227

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-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.

228

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-voice-account2] user telA password simple 12345

229

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-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

230

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-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


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


Default level

2: System level

Parameters



Examples

# Enable call forwarding busy for voice subscriber line 1/0 and set the forwarded-to number to 12345678.



[Sysname-subscriber-line1/0] call-forwarding on-busy enable forward-number 12345678

231

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:


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


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-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.



NOTE:


232

Syntax

call-forwarding unavailable enable forward-number number

undo call-forwarding unavailable enable

View


Default level

2: System level

Parameters



Examples

# Enable call forwarding unavailable for voice subscriber line 1/0 and set the forwarded-to number to

12345678.



[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.


unavailable enable, and call-forwarding priority.

NOTE:


Syntax

call-forwarding unconditional enable forward-number number

undo call-forwarding unconditional enable

View


Default level

2: System level

Parameters



233

Examples

# Enable call forwarding unconditional for voice subscriber line 1/0 and set the forwarded-to number to

12345678.



[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


Default level

2: System level

Parameters

None

Examples

# Enable the call hold feature for voice 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

234

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-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:


Call hold must be enabled before call transfer.

Syntax

call-transfer enable

undo call- transfer enable

View


Default level

2: System level

235

Parameters

None

Examples

# Enable call transfer for voice 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:


Call hold must be enabled before call transfer.

Syntax

call-transfer start-delay number

undo call-transfer start-delay

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-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.

236

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


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-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


Default level

2: System level

Parameters

None

Examples

# Enable call waiting for voice subscriber line 1/0.

237



[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


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-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.

238

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


Default level

2: System level

Parameters

None

Examples

# Enable the three-party conference function for voice 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


Default level

2: System level

Parameters

None

Examples

# Enable incoming call barring for voice subscriber line 1/0.


239


[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


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-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

240

Parameters







Examples

# Display the information of subscription.

<Sysname> display voice sip subscribe-state

Number Server Address Expires Status

+-----------------------------------------------------------------------+

1515 100.1.1.101:5060 3600 online


Field Description

Number Phone number that proposes the subscription

Server Address MWI server address, in the format of IP address plus port number or domain

name

Expires Effective time for the subscription

Status

Subscription state:

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.

241

number number: Displays the MWI information of a specified number.







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)


Field Description

MWI type

MWI types:

NoBind-S: Strict match non-binding.

NoBind-L: Loose match non-binding.

Bind: Binding

MWI server MWI server address, in the format of IP address plus port number or domain

name.

MWI expires Effective time for the subscription.

Number

Phone number.

As shown in the example, number: 1515 indicates that this is the MWI

information for number 1515.

Messages-Waiting

Message waiting identifier:

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.

242

Field Description

Voicemail

Number of new messages/number of old messages (number of new urgent

messages/ number of old urgent messages):

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

Total number of normal messages (total number of urgent messages).

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:


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


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-subscriber-line1/0] feature permit

243

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


Default level

2: System level

Parameters

None

Examples

# Enable hunt group for voice 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

244

undo hunt-group priority

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-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


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-subscriber-line1/0] joined-conference enable

245

mwi enable Description

Use mwi enable to enable MWI.

Use undo mwi enable to disable MWI.

By default, MWI is disabled.

NOTE:


Syntax

mwi enable

undo mwi enable

View


Default level

2: System level

Parameters

None

Examples

# Enable MWI for voice 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:


Syntax

mwi tone-duration length

undo mwi tone-duration

View


246

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-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.

247

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-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-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-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.

248

Syntax

timer called-hookon-delay seconds

undo timer called-hookon-delay

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-subscriber-line1/0] timer called-hookon-delay 90

249

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-E1 1/0] call-watch group 1 soft

250

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] call-watch rule 1 local interface ethernet 1/1

# Create monitor group 2 and associate it with track object ID 1.


[Sysname] call-watch rule 2 remote track 1

251

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 }


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.







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


Field Description

Controller E1

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.

Call-Watch mode The operating mode of the call-watch group on the interface, which can be

hard or soft.

Call-Watch rule 1 remote track

1

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.

252

Field Description

state is INVALID

The state of the track object, which can be POSITIVE, iNVALID, or

NEGATIVE.

In this example, it is INVALID.

Call-Watch rule 2 local interface

Ethernet1/1 Indicate that monitor group 2 monitors local interface Ethernet 1/1.

network is DOWN

The network layer state of the monitored local interfaces, which can be up or

down.

It is down in this example.

253

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 { g711alaw | g711ulaw }

default entity fax train-mode { local | ppp }

default entity fax cng-switch enable

default entity modem protocol pcm { standard | nte-compatible } { g711alaw | g711ulaw }

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


Default level

2: System level

254

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.


transmission rate to 9600 bps.


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.

g711alaw: Adopts G.711 A-law.

g711ulaw: Adopts G.711 μ-law.

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.

train-mode: Specifies the fax training mode. It defaults to ppp.

local: Adopts local training.

ppp: Adopts point-to-point training.

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-voice-dial] default entity fax baudrate 9600

255

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







Examples

# Display the statistics about the FoIP module.

<Sysname> display voice fax statistics

Statistics about Fax Session:

{

Total : 0

FAX_VOFR_STANDARD_SWITCH: 0

FAX_VOFR_FRF11_TRUNK : 0

FAX_VOFR_FRF11_SWITCH : 0

FAX_VOFR_MOTOROLA : 0

FAX_VOIP_STDT38 : 0

FAX_VOIP_T38 : 0

Success : 0





FAX_VOIP_STDT38 : 0

FAX_VOIP_T38 : 0

Failure : 0




256


FAX_VOIP_STDT38 : 0

FAX_VOIP_T38 : 0

Last Time : 00:00:00

FAX_VOFR_STANDARD_SWITCH: 00:00:00

FAX_VOFR_FRF11_TRUNK : 00:00:00

FAX_VOFR_FRF11_SWITCH : 00:00:00

FAX_VOFR_MOTOROLA : 00:00:00

FAX_VOIP_STDT38 : 00:00:00

FAX_VOIP_T38 : 00:00:00

Processed Pages : 0





FAX_VOIP_STDT38 : 0

FAX_VOIP_T38 : 0

}

Statistics about using fax baudrate:

{

V27 2400 : 0

V27 4800 : 0

V29 7200 : 0

V29 9600 : 0

V17 7200 : 0

V17 9600 : 0

V17 12000: 0

V17 14400: 0

}

Statistics about using ECM or Non-ECM mode:

{

ECM : 0

Non-ECM: 0

}

Statistics about release reason:

{

WAIT_DP_BEG_DEMODULATE_TIMEOUT : 0

WAIT_DP_BEG_MODULATE_TIMEOUT : 0

WAIT_DP_END_DEMODULATE_TIMEOUT : 0

WAIT_DP_END_MODULATE_TIMEOUT : 0

WAIT_FRAMEACK_TIMEOUT : 0

WAIT_T30MSG_PSTN_TIMEOUT : 0

WAIT_T30MSG_IP_TIMEOUT : 0

257

SPOOL_TIME_OVER : 0

GET_INVALID_T30MESSAGE : 0

IPP_CALL_RELEASE : 0

NORMAL_RELEASE : 0

UNKNOWN_REASON : 0

}


Field Description

FAX_VOFR_STANDARD_SWITCH Fax statistics for standard VoFR

FAX_VOFR_FRF11_TRUNK Fax statistics for FRF.11 trunk VoFR

FAX_VOFR_FRF11_SWITCH Fax statistics for FRF.11 switched VoFR

FAX_VOFR_MOTOROLA Fax statistics for Motorola compatible VoFR

FAX_VOIP_STDT38 Fax statistics for standard T.38 VoIP

FAX_VOIP_T38 Fax statistics for T.38 VoIP

WAIT_DP_BEG_DEMODULATE_TIMEOUT

Statistics of the number of connections released in the

case that the DP does not start demodulation within the

specified time

WAIT_DP_BEG_MODULATE_TIMEOUT


case that the DP does not start modulation within the

specified time

WAIT_DP_END_DEMODULATE_TIMEOUT


case that the DP does stop demodulation within the

specified time

WAIT_DP_END_MODULATE_TIMEOUT


case that the DP does not stop modulation within the

specified time

WAIT_FRAMEACK_TIMEOUT


case that no Frame ACK message is received from the DP

within the specified time

WAIT_T30MSG_PSTN_TIMEOUT


case that no T.30 message is received from PSTN within

the specified time

WAIT_T30MSG_IP_TIMEOUT


case that no T.30 message is received from the IP network

within the specified time

SPOOL_TIME_OVER


case that the number of spooling attempts exceeds the

maximum

GET_INVALID_T30MESSAGE Statistics of the number of connections released owing to

invalid T.30 message

IPP_CALL_RELEASE Statistics of the number of released IPP calls

NORMAL_RELEASE Statistics of the number of connections released normally

258

Field Description

UNKNOWN_REASON 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.


rate is 9,600 bps.


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.711 is adopted, the fax baud rate is 14,400 bps and the fax protocol is V.17.

If G.723.1 Annex A is adopted, the fax baud rate is 4,800 bps and the fax protocol is V.27.



Examples

# Configure the gateway to negotiate the fax rate in accordance with the V.29 fax protocol.





259

[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


Default level

2: System level

Parameters

None

Examples

# Enable the CNG fax switchover function.

<sysmane> system-view


[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

260

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-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


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-voice-dial-entity4] fax level -20

261

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


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-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

262

undo fax nsf-on

View


Default level

2: System level

Parameters

None

Examples

# Configure a nonstandard faculty for fax signal transmission.





[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.711 A-law and G.711 law are

supported, and the VAD function should be disabled.

Syntax

fax protocol { t38 | standard-t38 } [ hb-redundancy number | lb-redundancy number ]

fax protocol pcm { g711alaw | g711μlaw }

undo fax protocol

263

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.

g711alaw: Enables G.711 A-law.

g711ulaw: Enables G.711 μ-law.

Examples

# Set to 2 the number of high-speed redundant packets sent via the T.38 fax recommendation.





[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


Default level

2: System level

264

Parameters

local: Adopts the local training mode.

ppp: Adopts the ppp training mode.

Examples

# Configure the local training mode for the gateway.





[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


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-voice-dial-entity10] modem compatible-param 99

265

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 } { g711alaw | g711ulaw }

undo modem protocol

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.

g711alaw: Uses g711alaw codec for Modem pass-through.

g711ulaw: Uses g711ulaw codec for Modem pass-through.

Examples

# Set the switching mode to NTE-compatible and the codec type to g711alaw for SIP Modem pass-through.





[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

266

Examples

# Clear IP fax statistics.

<Sysname> reset voice fax statistics

267

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-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-voice-ivr-node1] call-normal length 7

# Configure node 1 to receive a normal secondary-call number by matching the number.




268


[Sysname-voice-ivr-node1] call-normal matching


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-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




269




Examples

# Display IVR call information.

<Sysname> display voice ivr call-info

Index Called-Number Caller-Number Entity Node-Id Status

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

1 101 100 101 1 PLAY MEDIA

2 406 200 201 3 WAIT INPUT

3 606 300 301 6 CALL

4 806 400 401 9 IDLE


Field Description

Index Index of the call information

Called-Number Number of the called party

Caller-Number Number of the calling party

Entity IVR voice entity number of the called number

Node-Id Node ID

Status

Current status:

IDLE: The node is idle.

PLAY MEDIA: The node is playing media files.

WAIT INPUT: The node is waiting for the input of the

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



270





Examples

# Display the IVR playing information.

<Sysname> display voice ivr media-play

Index Codec Media-Id Play-Times Status Type

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

1 g729r8 1001 3 play PSTN:1/0

2 g711alaw 1002 2 stop IP:100.1.1.1

3 g711ulaw 1003 2 stop IP:100.1.1.1

4 g723r53 1004 2 stop IP:100.1.1.1


Field Description

Index Playing index

Codec

Codec type, taking the values:

g729r8

g711alaw

g711ulaw

g723r53

Media-Id Media resource file ID

Play-Times Play times of a file

Status

Current status:

play

stop

Type

Current play type:

PSTN: The called party is from PSTN. In the example,

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

271

Default level

2: System level

Parameters







Examples

# Display IVR media resource information.

<Sysname> display voice ivr media-source

Codec Media-Id source Size (Bytes) Read-Number Cache-Number

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

g729r8 1000 cfa0:/wav/g7 69304 1 1

29r8/0.wav


Field Description

Codec Codec type of the media resource file

Media-Id Media resource file ID

Source Media source:

The file name is displayed if the media resource is a file.

Size (Bytes)

Size of the media resource, in bytes.

The size of the file is displayed if the media resource is a

file.

Read-Number Number of the read control block

Cache-Number 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 }

272

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.





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.




273


[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-voice-ivr-node1] input-error end-call media-play 1000 6 repeat 5


274

The node should return to the previous node after the maximum number of times permitted for inputting

errors is reached.


The node plays voice prompts only once.

The maximum number of input errors is three.





[Sysname-voice-ivr-node1] input-error goto-pre-node media-play 1001 1 repeat 3


The node should jump to node 20 after the maximum number of times permitted for inputting errors is

reached.


The node plays voice prompts three times.





[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.


275

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 node plays voice prompts twice.





[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-voice-dial-entity100] ivr-root 1

ivr-system Description

Use ivr-system to enter IVR management view.

276

Syntax

ivr-system

View

Voice view

Default level

2: System level

Parameters

Node

Examples

# Enter IVR management view.




[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.


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:

277

The timeout time is 20 seconds.


The node plays voice prompts only once.

The maximum number of timeout is twice




[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 { g711alaw | g711ulaw | g723r53 | g729r8 }

View

IVR management view

Default level

2: System level

Parameters

g711alaw: Enters g711alaw codec view.

g711ulaw: Enters g711ulaw codec view.

g723r53: Enters g723r53 codec view.

g729r8: Enters g729r8 codec view.

Examples

# Enter g729r8 codec view.





[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.

278

Syntax

media-play media-id [ play-times ] [ force ]

undo media-play

View


Default level

2: System level

Parameters

media-id: Media resource file ID, in the range 0 to 2147483647.


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-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.

279

all: All types of nodes.

Examples

# Create Jump node 1 and enter its view.





[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.


Examples

# Configure Service node 1.



280


[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-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.

281

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-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


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.

282

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.


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-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.


283




[Sysname-voice-ivr-node1] user-input 0 end-call

284

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.11 sub-channel number to be configured must be available; the FRF.11 sub-channel is not occupied.

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.11 trunk mode.

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.11 sub-channel number of a VoFR entity, in the range of 4 to 255.

Examples

# Specify DLCI 100 to adopt the dynamic call mode.




[Sysname-voice-dial] entity 4 vofr


[Sysname-voice-dial-entity4] address vofr-dynamic serial1/0 100

285

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.11

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.11 sub-channel. The call control protocol

used in the dynamic call mode is specified by executing vofr in interface DLCI view.

FRF.11 trunk mode: A voice channel is established when you execute the address vofr-static command.

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.11 trunk mode, you must use trunk-id to configure a

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.11 trunk mode.

Examples

# Configure the FRF.11 trunk mode for VoFR entity 10.





[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.

286

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] 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.11 sub-channel information on a VoFR DLCI. You can use the

display fr vofr-info serial interface-number command to display the FRF.11 sub-channel information on a

specified interface sub-interface. The information of all FRF.11 sub-channels will be displayed if no interface

sub-interface is specified. You can use the display fr vofr-info dlci-number to display the FRF.11 sub-channel

information on a specified DLCI.

Syntax

display fr vofr-info [ serial interface-number [ dlci-number ] ] [ | { begin | exclude | include }


View

Any view

Default level

2: Monitor level

Parameters

serial interface-number: Displays the FRF.11 sub-channel information on a specified interface.

dlci-number: Virtual circuit number, in the range of 16 to 1007



287





Examples

# Display the FRF.11 sub-channel information on a VoFR DLCI.

<Sysname> display fr vofr-info

interface(dlci) vofr-mode cid cid-type

Serial2/0:0(100) vofr-nonstandard 5 frag-data

Serial2/0:0(100) vofr-nonstandard 4 voice-signal

# Display the FRF.11 sub-channel information on the specified interface.

<Sysname> display fr vofr-info ser2/0:0 17

interface(dlci) vofr-mode cid cid-type pkts(in/out/drop)

Serial2/0:0(17) vofr-nonstandard 254 frag-data 0/0/0

Serial2/0:0(17) vofr-nonstandard 255 voice-signal 0/1068/0


Field Description

interface(dlci) Frame relay interface name (DLCI number)

vofr-mode VoFR call control protocol, for example, VoFR nonstandard-compatible and

VoFR-Huawei-compatible.

cid Voice channel number

cid-type Type of a voice channel

pkts(in/out/drop) 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


Default level

2: System level

288

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.





outband vofr Description

Use outband vofr to configure the out-of-band DTMF transmission mode.

Use undo outband to restore the default.


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-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.

289

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-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

290

Parameters

None

Examples

# Configure voice packets sent by VoFR entity 10 to carry a timestamp.





[Sysname-voice-dial-entity10] timestamp

trunk-id Description

Use trunk-id to configure a PSTN-dialed number in the FRF.11 trunk mode.

Use undo trunk-id to restore the default.

By default, no PSTN-dialed number is configured in the FRF.11 trunk mode.

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.11 trunk mode.





[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.

291

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] 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.11 trunk mode),

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.


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

292

Parameters

huawei-compatible: Adopts the Huawei-compatible mode.

motorola-compatible: Adopts the Motorola-compatible mode for compatibility with VoFR of Motorola

routers. The FRF.11 trunk mode does not support the Motorola-compatible protocol.

dce: Specifies the virtual circuit to serve as a DCE in compliance with Annex G.

dte: Specifies the virtual circuit to serve as a DTE in compliance with Annex G.

nonstandard-compatible: Adopts the nonstandard-compatible mode for compatibility with VoFR of Cisco

routers.

signal-channel ccid-no data-channel dcid-no: FRF.11 sub-channel numbers respectively used by signaling

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-Serial1/0] link-protocol fr ietf


[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-Serial1/0] link-protocol fr ietf


[Sysname-fr-dlci-Serial1/0-100] vofr huawei-compatible dte

vofr frf11-timer Description

Use vofr frf11-timer to configure the trunk wait timer length in the FRF.11 trunk mode.

Use undo vofr frf11-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.11 trunk calls after the command

is executed.


293

NOTE:

The Trunk Wait timer is specific to the FRF.11 trunk mode. Within the trunk wait timer length, incoming calls are

prohibited and received voice packets are dropped.

No signaling is exchanged in the FRF.11 trunk mode. When one voice gateway receives the first voice packet from

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.11 trunk

mode.

Syntax

vofr frf11-timer time

undo vofr frf11-timer

View

Voice view

Default level

2: System level

Parameters

time: Trunk Wait timer length in the FRF.11 trunk mode in seconds, in the range of 10 to 600.

Examples

# Configure the Trunk Wait timer length in the FRF.11 trunk mode to 40 seconds.



[Sysname-voice] vofr frf11-timer 40

294

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-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

295

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-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.


296



[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


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-voice-aaa] acct-method start-ack

# Restore the default accounting method.

[Sysname-voice-aaa] undo acct-method

297

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-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.

298

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


Default level

2: System level

Parameters

None

Examples

# Enable the authentication function for one-stage dialing users.




[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

299

Default level

2: System level

Parameters

None

Examples

# Enable the authorization function for users who dial the 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


Default level

2: System level

Parameters

None

Examples

# Enable the authorization function for one-stage dialing users.



300


[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-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

301

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-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.

302

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


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-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

303

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







Examples

# Display the configuration information and access numbers.

<Sysname> display voice access-number

AAA configuration :

accounting-method = start-ack

accounting-did = on

authentication-did = off

authorization-did = on

call history rule:

cdr buffer = 100

cdr duration = 86400

cdr threshold = 50

access number: [ 17909 ]

dialing process = cardnumber

accounting = on

authentication = on

authorization = on

callednum receive = termintor

card digit = 12

password digit = 6

redialing times = 2

access number: [ 201 ]

dialing process = voice-caller

accounting = off

authentication = off

authorization = off

304

callednum receive = immediate

redialing times = 2

language selected = Chinese


Field Description

accounting-method Accounting method, including start-ack, start-no-ack, and stop-only.

See the acct-method command.

accounting-did

Accounting function for one-stage dialing users.

on: Enabled.

off: Disabled.

See the accounting-did command.

authentication-did

Authentication function for one-stage dialing users.

on: Enabled.

off: Disabled.

See the authentication-did command.

authorization-did

Authorization function for one-stage dialing users.

on: Enabled.

off: Disabled.

See the authorization-did command.

call history rule Rule for saving CDRs.

cdr buffer Number of CDRs that can be saved.

See the cdr buffer command.

cdr duration Lifetime of CDRs. See the cdr duration command.

cdr threshold CDR alarm threshold.

See the cdr threshold command.

access number Access number, for example, 17909.

See the gw-access-number command.

dialing process

Two-stage dialing process, including card number/password process, caller

number process, caller number process with IVR.

See the process-config command.

accounting

Accounting function for two-stage dialing users.

on: Enabled.

off: Disabled.

See the accounting command.

authentication

Authentication function for two-stage dialing users.

on: Enabled.

off: Disabled.

See the authentication command.

305

Field Description

authorization

Authorization function for two-stage dialing users

on: Enabled

off: Disabled

See the authorization command.

callednum receive

Method of collecting digits of a called number, including terminator and

immediate

See the callednumber receive-method command.

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 callee‘s 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

306

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 callee‘s IP address. The ip-address argument represents

a callee‘s IP address.







Examples

# Display call records by calling number.

<Sysname> display voice call-history-record callingnumber 4000

Call records of voice RADIUS:

#

CallRecord [ 0 ]:

CallReference = 46

CallRecordTime = Oct 20, 2006 16:45:47

CardNumber = None

AccessNumber = None

Incoming call leg:

CallingNumber = 4000

SignalType = FXS/O

VoiceInterface = 1/0

SetupTime = Oct 20, 2006 16:45:43

ConnectTime = Oct 20, 2006 16:45:45

ReleaseTime = Oct 20, 2006 16:45:47

SendPackets = 71 packages

SendBytes = 2982 bytes

ReceivePackets = 111 packages

ReceiveBytes = 4662 bytes

Outgoing call leg [ 0 ]:

307

CalledNumber = 2000

CallDuration = 00h 00m 02s

EncodeType = G729R8

DecodeType = G729R8

ReleaseCause = Called hook on

SignalType = SIP

IpAddress/Port = 1.1.1.19/5060

SetupTime = Oct 20, 2006 16:45:43

ConnectTime = Oct 20, 2006 16:45:45

ReleaseTime = Oct 20, 2006 16:45:47

SendPackets = 111 packages

SendBytes = 4662 bytes

ReceivePackets = 72 packages

ReceiveBytes = 3024 bytes

#

The end


Field Description

Call records of voice RADIUS Voice RADIUS call records

CallRecord [ 0 ] Call record number

CallReference Voice RADIUS module call identification

CallRecordTime Time when a call is recorded

CardNumber Card number

AccessNumber Access number

Incoming Call Leg Information of the incoming call leg

CallingNumber Calling number

SignalType Signaling protocol type (for example, R2, E&M)

VoiceInterface Voice interface

SetupTime Call setup time

ConnectTime Call-connected time

ReleaseTime Call release time

SendPackets Packets sent

SendBytes Bytes sent

ReceivePackets Packets received

ReceiveBytes Bytes received

Outgoing call leg [ 0 ] Information of the outgoing call leg. One call may involves multiple

outgoing call legs. [ 0 ] identifies one outgoing call leg.

CalledNumber Called number

CallDuration Call duration

EncodeType Encoding type

308

Field Description

DecodeType Decoding type

ReleaseCause Call release cause

SignalType Signaling protocol (for example, R2, E&M)

VoiceInterface Voice interface

IpAddress/Port IP address and port number

SetupTime Call setup time

ConnectTime Call-connected time

ReleaseTime Call release time

SendPackets Packets sent

SendBytes Bytes sent

ReceivePackets Packets received

ReceiveBytes 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







Examples

# Display statistics of messages exchanged between the voice RADIUS module, CMC module, and AAA

module.

<Sysname> display voice radius statistic

VORDS => AAA:

309

Authen_Request = 0

Author_Request = 0

AcctReq_PstnCaller = 0

AcctReq_VoipCaller = 0

AcctReq_PstnCalled = 0

AcctReq_VoipCalled = 0

Account_Stop = 0

Leaving = 0

AAA => VORDS:

Authen_Accept = 0

Authen_Reject = 0

Author_Accept = 0

Author_Reject = 0

AcctRsp_PstnCaller = 0

AcctRsp_VoipCaller = 0

AcctRsp_PstnCalled = 0

AcctRsp_VoipCalled = 0

Account_Ok = 0

Account_Failure = 0

Cut = 0

CMC => VORDS:

Setup = 0

Alerting = 0

Connect = 0

Release = 0

DtmfInformation = 0

ChannelReady = 0

FaxVoiceSwitch = 0

FaxTone = 0


Field Description

VORDS=>AAA: Messages from the voice RADIUS module to the AAA module

Authen_Request Authentication_Request message

Author_Request Authorization_Request message

AcctReq_PstnCaller Accounting_Request message for PSTN caller

AcctReq_VoipCaller Accounting_Request message for VoIP caller

AcctReq_PstnCalled Accounting_Request message for PSTN callee

AcctReq_VoipCalled Accounting_Request message for VoIP callee

Account_Stop Accounting_Stop message

Leaving Leaving message

AAA=>VORDS: Messages from the AAA module to the voice RADIUS module

Authen_Accept Authentication_Accept message

Authen_Reject Authentication_Reject message

310

Field Description

Author_Accept Authorization_Accept message

Author_Reject Authorization_Reject message

AcctRsp_PstnCaller Accounting_Response message for PSTN caller

AcctRsp_VoipCaller Accounting_Response message for VoIP caller

AcctRsp_PstnCalled Accounting_Response message for PSTN callee

AcctRsp_VoipCalled Accounting_Response message for VoIP callee

Account_Ok Accounting_Ok message

Account_Failure Accounting_Failure message

Cut Cut message

CMC=>VORDS: Messages from the CMC module to the voice RADIUS module

Setup Setup message

Alerting Alerting message

Connect Connect message

Release Release message

DtmfInformation DTMF digit

ChannelReady Channel_Ready message

FaxVoiceSwitch Fax_Voice_Switch message

FaxTone 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 }

311

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-voice-dial-anum17909]

# Add the access number 179 and enter access number view.




[Sysname-voice-dial] gw-access-number 179..

[Sysname-voice-dial-anum179..]

# Delete the access number 17909.




[Sysname-voice-dial] undo gw-access-number 17909

# Delete all access numbers.




[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.

312

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.


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-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.

313

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.






# 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.

314

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.


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-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

315

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.


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-voice-dial-anum17909] process-config voice-caller

[Sysname-voice-dial-anum17909] selectlanguage english

316

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-voice-dial-anum17909] timer two-stage dial-interval 5

317

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

http://www.hp.com/support

http://www.hp.com/go/wwalerts


http://www.hp.com/

http://www.hp.com/go/networking


http://www.hp.com/support/downloads

http://www.software.hp.com/

318

Conventions This section describes the conventions used in this documentation set.

Command conventions

Convention Description

Boldface Bold text represents commands and keywords that you enter literally as shown.

Italic Italic text represents arguments that you replace with actual values.

[ ] Square brackets enclose syntax choices (keywords or arguments) that are optional.

{ x | y | ... } Braces enclose a set of required syntax choices separated by vertical bars, from which

you select one.

[ x | y | ... ] Square brackets enclose a set of optional syntax choices separated by vertical bars, from

which you select one or none.

{ x | y | ... } * Asterisk-marked braces enclose a set of required syntax choices separated by vertical

bars, from which you select at least one.

[ x | y | ... ] * Asterisk-marked square brackets enclose optional syntax choices separated by vertical

bars, from which you select one choice, multiple choices, or none.

&<1-n> 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


Boldface 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


WARNING An alert that calls attention to important information that if not understood or followed can

result in personal injury.

CAUTION 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.

IMPORTANT An alert that calls attention to essential information.

NOTE An alert that contains additional or supplementary information.

TIP An alert that provides helpful information.

319

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.

320

Index

A B C D E F G H I J K L M N O P Q R S T U V W A

aaa-client,294

account enable,213

accounting,294

accounting-did,295

acct-method,296

address,211

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

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

321

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,11

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,11

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

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

322

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

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,311

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

323

proxy server-group,223

Q

qsig-tunnel enable,110

R

re-answer enable,110

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,111

registrar,179

registrar server-group,223

remote-party-id,181

renew,113

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,114

ring-detect debounce,80

ring-detect frequency,81

rtp payload-type nte,38

rule,183

rule,143

rule,205

S

seizure-ack enable,115

selectlanguage,315

select-mode,115

select-rule operation-order,280

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,116

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,117

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,118

srs,199

subscriber-group,151

subscriber-line,84

subscriber-line,119

substitute (voice dial program view),153

substitute (voice subscriber line view, voice entity

view),152

T

tdm-clock,119

terminator,154

timeout,281

timer called-hookon-delay,247

timer connection age,188

timer dial-interval,84

timer disconnect-pulse,85

324

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 frf11-timer,292

voice bandwidth,290

voice-setup,41

voip timer,42

vqa dscp,42

vqa dsp-monitor buffer-time,44

W

wildcard-register enable,195