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
Support and other resources······································································································································ 317 Contacting HP ······························································································································································ 317
Subscription service ············································································································································ 317 Related information ······················································································································································ 317
Documents ···························································································································································· 317
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Websites ······························································································································································ 317 Conventions ·································································································································································· 318
Index ············································································································································································· 320
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
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
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
Table 4 G.723 r53 algorithm
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
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.
Table 5 G.726 r16 algorithm
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 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.
Table 6 G.726 r24 algorithm
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 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
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
70 ms 210 250 28.6 kbps 256 29.3 kbps 70 ms
G.726 r24 algorithm: media stream bandwidth 24 kbps, minimum packet assembly interval 10 ms.
Table 7 G.726 r32 algorithm
Packet assembly interval
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 40 80 64 kbps 86 68.8 kbps 10 ms
20 ms 80 120 48 kbps 126 50.4 kbps 20 ms
30 ms 120 160 42.7 kbps 166 44.3 kbps 30 ms
40 ms 160 200 40 kbps 206 41.2 kbps 40 ms
50 ms 200 240 38.4 kbps 246 39.4 kbps 50 ms
G.726 r32 algorithm: media stream bandwidth 32 kbps, minimum packet assembly interval 10 ms.
Table 8 G.726 r40 algorithm
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 50 90 72 kbps 96 76.8 kbps 10 ms
20 ms 100 140 56 kbps 146 58.4 kbps 20 ms
30 ms 150 190 50.7 kbps 196 52.3 kbps 30 ms
40 ms 200 240 48 kbps 246 49.2 kbps 40 ms
G.726 r40 algorithm: media stream bandwidth 40 kbps, minimum packet assembly interval 10 ms.
Table 9 G.729 algorithm
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 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
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
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
Bytes coded in a time unit
Before compression After IPHC+PPP compression
Packet length (IP+PPP) (bytes)
Network bandwidth (IP+PPP)
Packet length (IP+PPP) (bytes)
Network bandwidth (IP+PPP)
G.729 30 76 20.3 kbps 34 9.1 kbps
G.723r63 24 70 18.4 kbps 28 7.4 kbps
G.723r53 20 66 17.5 kbps 24 6.4 kbps
G.726r16 60 106 28.3 kbps 64 17.1 kbps
6
Codec
Bytes coded in a time unit
Before compression After IPHC+PPP compression
Packet length (IP+PPP) (bytes)
Network bandwidth (IP+PPP)
Packet length (IP+PPP) (bytes)
Network bandwidth (IP+PPP)
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.
g723r63: G.723.1 Annex A codec, requiring a bandwidth of 6.3 kbps.
g726r16: G.726 Annex A codec. It uses the ADPCM technology, requiring a bandwidth of 16 kbps.
g726r24: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 24 kbps.
7
g726r32: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 32 kbps.
g726r40: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 40 kbps.
g729a: G.729 Annex A codec (a simplified version of G.729), requiring a bandwidth of 8 kbps.
g729br8: G.729 Annex B codec. It uses CS-ACELP, requiring a bandwidth of 8 kbps.
g729r8: G.729 (the voice compression technology using conjugate algebraic-code-excited linear-prediction),
requiring a bandwidth of 8 kbps.
Examples
# Configure to use g723r53 coding/decoding algorithm first, then the g729r8.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] compression 1st-level g723r53
[Sysname-voice-dial-entity10] compression 2nd-level g729r8
default entity compression Description
Use default entity compression to specify the default global codecs and their priority levels.
Use undo default entity compression to restore the default.
By default, the codec with the first priority is g729r8, the codec with the second priority is 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.
g723r53: G.723.1 Annex A codec, requiring a bandwidth of 5.3 kbps.
g723r63: G.723.1 Annex A codec, requiring a bandwidth of 6.3 kbps.
g726r16: G.726 Annex A codec. It uses the ADPCM technology, requiring a bandwidth of 16 kbps.
g726r24: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 24 kbps.
g726r32: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 32 kbps.
g726r40: G.726 Annex A codec. It uses ADPCM, requiring a bandwidth of 40 kbps.
g729a: G.729 Annex A codec (a simplified version of G.729), requiring a bandwidth of 8 kbps.
g729br8: G.729 Annex B codec. It uses CS-ACELP, requiring a bandwidth of 8 kbps.
g729r8: G.729 (the voice compression technology using conjugate algebraic-code-excited linear-prediction),
requiring a bandwidth of 8 kbps.
Examples
# Adopt the g723r53 coding and decoding mode as the first selection globally.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] default entity compression 1st-level g723r53
default entity payload-size Description
Use default entity payload-size to configure the default packetization period for a codec.
Use undo default entity payload-size to restore the default.
Because the IVR voice entity does not support g726 codecs, the packetization periods configured for g726
codecs on an IVR voice entity take no effect. For more information about the IVR voice entity, see Voice
Configuration Guide.
Related commands: default entity compression, entity compression, payload-size, and set-media.
Syntax
default entity payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } time-length
undo default entity payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 }
View
Voice dial program 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.
g726r24: Specifies the packetization period for g726r24 codec. It ranges from 10 to 70 milliseconds and
defaults to 30 milliseconds.
g726r32: Specifies the packetization period for g726r32 codec. It ranges from 10 to 50 milliseconds and
defaults to 30 milliseconds.
g726r40: Specifies the packetization period for g726r40 codec. It ranges from 10 to 40 milliseconds and
defaults to 30 milliseconds.
g729: Specifies the packetization period for g729 codec. It ranges from 10 to 180 milliseconds and defaults
to 30 milliseconds.
time-length: Packetization period for a codec.
Examples
# Set the packetization period for G.711 codec to 30 milliseconds.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] default entity payload-size g711 30
default entity vad-on Description
Use default entity vad-on to globally configure VAD as the default value.
Use undo default entity vad-on to restore the fixed value (disabling VAD) to be the default value.
By default, VAD is disabled.
The default entity vad-on command is used to globally enable VAD and make it as the default setting. After
the configuration, all the voice entities and newly created voice entities on this router, which have not been
configured with this function, will inherit this configuration (G. 711 does not support VAD).
Related commands: vad-on.
Syntax
default entity vad-on
undo default entity vad-on
View
Voice dial program view
Default level
2: System level
Parameters
None
Examples
# Enable VAD globally.
10
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] default entity vad-on
description (voice entity view) Description
Use description to configure a voice entity description string.
Use undo description to delete the voice entity description string.
By default, no description is configured for the voice entity.
You can use description to add a description to a voice entity, which has no effect on the performance of the
voice entity interface. You can view this description with the display command.
Syntax
description string
undo description
View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view
Default level
2: System level
Parameters
string: Voice entity description string, whose length ranges from 1 to 80 characters.
Examples
# Add the description local-entity 10 to voice entity 10.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] description local-entity10
dial-trap enable Description
Use dial-trap enable to enable the trap function for a voice entity.
Use undo dial-trap enable to disable the trap function for a voice entity.
By default, the trap function is disabled for a voice entity.
Syntax
dial-trap enable
undo dial-trap enable
View
POTS voice entity view, VOIP voice entity view, VoFR entity view, IVR entity view
11
Default level
2: System level
Parameters
None
Examples
# Enable the trap function for VoIP voice entity 10.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] dial-trap enable
dial-program Description
Use dial-program to enter the voice dial program view.
Syntax
dial-program
View
Voice view
Default level
2: System level
Parameters
None
Examples
# Enter the dial program view
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
display voice call-info Description
Use display voice call-info to display the contents in the call information table.
Syntax
display voice call-info { brief | mark tag | verbose } [ | { begin | exclude | include } regular-expression ]
View
Any view
Default level
2: System level
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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the information of the call control block of the CMC module.
<Sysname> display voice cmc ccb
The CMC Module Call Control Block Information!
#
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
}
Table 12 Output description
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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the current default values and the system-default values.
<Sysname> display voice default all
default entity fax ecm off(system: off)
default entity fax protocol t38(system: t38)
default entity fax protocol t38 hb-redundancy 0(system: 0)
default entity fax protocol t38 lb-redundancy 0(system: 0)
default entity fax level -10(system: -15)
default entity fax local-train threshold 10(system: 10)
default entity fax baudrate voice(system: voice)
default entity fax nsf-on off(system: off)
default entity fax train-mode ppp(system: ppp)
default entity fax cng-switch off(system: off)
default entity compression 1st-level g729r8(system: g729r8)
default entity compression 2nd-level g711alaw(system: g711alaw)
default entity compression 3rd-level g711ulaw(system: g711ulaw)
default entity compression 4th-level g723r53(system: g723r53)
default entity vad-on off(system: off)
default entity payload-size g711 20(system: 20)
default entity payload-size g723 30(system: 30)
default entity payload-size g726r16 30(system: 30)
default entity payload-size g726r24 30(system: 30)
default entity payload-size g726r32 30(system: 30)
default entity payload-size g726r40 30(system: 30)
default entity payload-size g729 30(system: 30)
17
default entity modem compatible-param 100(system: 100)
default entity modem protocol pcm disable
Table 13 Output description
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 g723 Voice entity packet assembly interval (G.723)
payload-size g726r16 Voice entity packet assembly interval (G.723 r16)
payload-size g726r24 Voice entity packet assembly interval (G.723 r24)
payload-size g726r32 Voice entity packet assembly interval (G.723 r32)
payload-size g726r40 Voice entity packet assembly interval (G.723 r40)
payload-size g729 Voice entity packet assembly interval (G.729)
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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the configuration information of POTS voice entities.
<Sysname> display voice entity all
Current configuration of entities
#
entity 100 pots
line 8/0
match-template 1000
#
End
Table 14 Output description
Field 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 }
regular-expression ]
View
Any view
Default level
2: System level
Parameters
all: Displays all statistics about the IPP module.
cmc: Displays statistics about the CMC module.
h225: Displays statistics about H.225 messages.
h245: Displays statistics about H.245 messages.
ras: Displays statistics about RAS messages.
socket: Displays statistics about socket messages.
timer: Displays timeout statistics.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display statistics about H.225 messages of the IPP module.
<Sysname> display voice ipp statistic h225
Statistics about H225 :
{
Send_Setup : 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
}
Table 15 Output description
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 }
regular-expression ]
View
Any view
Default level
2: System level
Parameters
all: Displays all the statistic information related to the IVA module.
call: Displays the calling statistics in the IVA module.
cmc: Displays all the interaction statistics between the IVA and the CMC module.
error: Displays all the error statistics of the IVA module.
isdn: Displays the interaction statistics between IVA module and ISDN.
proc: Displays the statistic information of process call in the IVA module.
timer: Displays the timer‘s statistic information of the IVA module.
vim: Displays all the interaction statistic information between IVA module and VIM.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the call statistics between IVA module and other modules.
<Sysname> display voice iva statistic call
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
}
Table 16 Output description
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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
23
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the statistics of all active calls.
<Sysname> display voice statistics call-active all
Current information of call active table:
General Info:
SetupTime:647449 ms
Index:1
PhoneNumber:200
PhoneSubNumber:
EntityIndex:
IfIndex:0x0
ConnectTime:673269 ms
CallDuration: 0 days 22h:49m:27s
CallState:Active
CallOrigin:Answer
ChargedUnits:0
CallInfoType:speech
ByteReceived:115070004
ByteTransmitted:115067526
PacketReceived:2739762
PacketTransmitted:2739703
VOIP Info:
ConnectionId:0x0013
CallId:0
RemoteSignallingIPAddress:100.1.1.224
RemoteSignallingPort:5060
RemoteMediaIPAddress:100.1.1.224
RemoteMediaPort:16420
VADSwitch:0
SessionProtocol:Sipv2
CodecType:G729r8
CallingNumber:200
CalledNumber:100
SubstCallingNumber:200
SubstCalledNumber:100
General Info:
SetupTime:647452 ms
Index:1
PhoneNumber:100
PhoneSubNumber:
EntityIndex:100
24
IfIndex:0x2c00c0
ConnectTime:673267 ms
CallDuration: 0 days 22h:49m:27s
CallState:Active
CallOrigin:Originate
ChargedUnits:0
CallInfoType:Speech
ByteReceived:115068030
ByteTransmitted:115067484
PacketReceived:2739715
PacketTransmitted:2739702
PSTN Info:
ConnectionId:0x0013
CallId:1
TxDuration:82191625 ms
VoiceTxDuration:82191060 ms
FaxTxDuration:0 ms
ImgPages:0
CodecType:G729r8
CallingNumber:200
CalledNumber:100
SubstCallingNumber:200
SubstCalledNumber:100
End
Table 17 Output description
Field 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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the history records of all calls that have ended.
<Sysname>display voice statistics call-history all
Current information of call history table:
Call-History Info:
Index:1
SetupTime:155451 ms
PhoneNumber:7001
EntityIndex:7001
IfIndex:0x2c00f0
ConnectTime:168010 ms
TerminateTime:171130 ms
CallOrigin:Originate
ChargedUnits:0
CallInfoType:Speech
ByteReceived:18816
ByteTransmited:18816
PacketReceived:448
PacketTransmited:448
PSTN Info:
ConnectionId:0x0000
CallId:1
TxDuration:65836 ms
VoiceTxDuration:25280 ms
FaxTxDuration:0 ms
27
ImgPages:0
CodecType:G729r8
CallingNumber:6001
CalledNumber:7001
SubstCallingNumber:6001
SubstCalledNumber:7001
Call-History Info:
Index:2
SetupTime:155448 ms
PhoneNumber:6001
EntityIndex:6000
IfIndex:0x0
ConnectTime:168011 ms
TerminateTime:171131 ms
CallOrigin:Answer
ChargedUnits:0
CallInfoType:Speech
ByteReceived:21798
ByteTransmited:18816
PacketReceived:519
PacketTransmited:448
VOIP Info:
ConnectionId:0x0000
CallId:0
RemoteSignallingIPAddress: 100.1.1.223
RemoteSignallingPort:5060
RemoteMediaIPAddress:100.1.1.223
RemoteMediaPort:16428
VADSwitch:0
SessionProtocol:Sipv2
CodecType:G729r8
CallingNumber:6001
CalledNumber:7001
SubstCallingNumber:6001
SubstCalledNumber:7001
End
Table 18 Output description
Field 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
ConnectTime Accumulated connect time to the peer since the system started
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.
For multiple times of fax, the values are added.
ImgPages Number of pages faxed.
For multiple times of fax, the values are added.
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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the call statistics of all voice entities.
<Sysname> display voice statistics entity all
Current statistics of all entities:
Index:100
Type:pots
Match-Template:100
ConnectTime:0 s
SuccessfulCalls:0
FailedCalls:0
AcceptedCalls:0
RefusedCalls:0
LastSetupTime:0 ms
Index:200
Type:pots
Match-Template:200
ConnectTime:758 s
SuccessfulCalls:0
FailedCalls:0
AcceptedCalls:1
RefusedCalls:0
LastSetupTime:6190ms
End
30
Table 19 Output description
Field Description
Index Entity index
Type Entity type, which can be POTS, VoIP, or Other.
Match-Template Number template
ConnectTime Accumulated connect time to the peer since the system started
up, in milliseconds
LastSetupTime Setup time of the last call, in milliseconds
distinguish-localtalk Description
Use distinguish-localtalk to enable the local call identification function.
Use undo distinguish-localtalk to disable this function.
By default, the local call identification function is disabled.
NOTE:
Configuring the three-party conference service in voice subscriber line view will invalidate the
configuration of the distinguish-localtalk command. For more information about the three-party
conference service, see Call services configuration commands.
Syntax
distinguish-localtalk
undo distinguish-localtalk
View
Voice view
Default level
2: System level
Parameters
None
Examples
# Enable the local call identification function.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] distinguish-localtalk
dscp media Description
Use dscp media to set the DSCP value in the ToS field in the IP packets that carry the RTP stream of the voice
entity.
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
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view
Default level
2: System level
Parameters
dscp-value: DSCP value in the range of 0 to 63 or the keyword 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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 2 voip
[Sysname-voice-dial-entity2] dscp media af41
entity Description
Use entity to enter voice entity view, or configure a voice entity and then enter its view if the voice entity does
not exist.
Use undo entity to remove the existing voice entity.
In a global view, use entity to enter a voice entity view, and use quit to return to the dial program view.
For more information about IVR and VoFR voice entities, see Voice Configuration Guide.
Related commands: line.
NOTE:
The entity-number assigned to a VoIP, POTS, or IVR entity must be unique among all VoIP and POTS entities.
The system supports up to 1,000 voice entities.
Syntax
entity entity-number [ pots | voip ]
undo entity { entity-number | all | pots | voip }
View
Voice dial program view
Default level
2: System level
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.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
line Description
Use line to associate the voice entity with a specified voice subscriber line.
Use undo line to remove this association.
By default, there is no association between a voice entity and a voice subscriber line.
Syntax
line line-number
undo line
View
POTS voice entity view
Default level
2: System level
Parameters
line-number: Number of a subscriber line.
Examples
# Associate voice entity 10 and voice subscriber line 1/0.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] line 1/0
match-template Description
Use match-template to configure the number template for a voice entity.
Use undo match-template to remove the configuration.
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
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view
Default level
2: System level
Parameters
match-string: Number template. Its format is [ + ] { string [ T ] [ $ ] | T }, with the maximum length of 31
characters. The characters are described in the following.
+: The plus sign itself does not have special meanings. It only indicates that the following string is an
effective number and the number is E.164-compliant.
$: Is the last character, indicating the end of the number. That means the entire called number must
match the string part before ―$‖.
T: Timer. It means the system is waiting the subscriber for dialing any number till: the number length
threshold is exceeded, or the subscriber inputs the terminator; or the timer expires. T is used to match
a number with any digits.
string: A string composed of any characters of ―0123456789#*.!+%[]() -‖. The meanings of the
characters are described in the following table:
Table 20 Meanings of the characters in string
Character 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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] match-template 5557922
# Configure a match template for VoIP voice entity 10.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 010 voip
[Sysname-voice-dial-entity10] match-template 5557922
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] address sip ip 10.1.1.2
[Sysname-voice-dial-entity10] outband sip
payload-size Description
Use payload-size to configure the voice packetization period for different codecs.
Use undo payload-size to restore the default.
By default, the voice packetization period for g971 is 20 milliseconds, and that for g723, g726, and g726
is 30 milliseconds.
Because the IVR voice entity does not support g726 codecs, the packetization periods configured for g726
codecs on an IVR voice entity take no effect. For more information about the IVR voice entity, see Voice
Configuration Guide.
Related commands: default entity compression, default entity payload-size, entity compression, and
set-media.
Syntax
payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 } time-length
undo payload-size { g711 | g723 | g726r16 | g726r24 | g726r32 | g726r40 | g729 }
36
View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity 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
range of 30 to 180, with a default of 30.
g726r16: Packetization period in milliseconds for g726r16 codec, an integral multiple of 10 in the range of
10 to 110, with a default of 30.
g726r24: Packetization period in milliseconds for g726r24 codec, an integral multiple of 10 in the range of
10 to 70, with a default of 30.
g726r32: Packetization period in milliseconds for g726r32 codec, an integral multiple of 10 in the range of
10 to 50, with a default of 30.
g726r40: Packetization period in milliseconds for g726r40 codec, an integral multiple of 10 in the range of
10 to 40, with a default of 30.
g729: Packetization period in milliseconds for g729r8 or g729a codec, an integral multiple of 10 in the
range of 20 to 180, with a default of 30.
time-length: DSP packetization period for a codec.
Examples
# Set the voice packetization period of the DSP for g711 codec to 30 milliseconds.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] payload-size g711 30
register-number Description
Use register-number to enable the VoIP gateway to register numbers of a voice entity with an SIP server.
Use undo register-number to disable a gateway from registering numbers of a voice entity with an SIP server.
By default, after configured with SIP-registration related parameters, a POTS voice entity initiates registration
to the SIP server.
In some cases, you need to configure the same POTS voice entity on multiple gateways. As a SIP server
cannot have the same number, you cannot register a POTS voice entity with a SIP server at the same time.
In other cases, you may need to register only some port numbers on the gateway with a SIP server to meet
some special requirements. You can use undo register-number to specify the voice entity whose number does
not need to be registered.
Related commands: match-template.
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] undo register-number
reset voice cmc statistic Description
Use reset voice cmc statistic to clear calling statistics on the CMC module.
Related commands: display voice cmc.
Syntax
reset voice cmc statistic
View
User view
Default level
2: System level
Parameters
None
Examples
# Clear calling statistics on the CMC module.
<Sysname> reset voice cmc statistic
reset voice ipp statistic Description
Use reset voice ipp statistic to reset IPP statistics.
Related commands: display voice ipp statistic.
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] rtp payload-type nte 102
send-ring Description
Use send-ring to enable the local end to play ringback tone.
Use undo send-ring to disable the local end from playing ringback tone.
By default, the local end does not play ringback tone.
In VoIP view, this command is available only after the fast connection function is enabled or a SIP routing
policy is configured. In POTS view, you can configure this command as long as the line line number
command binds the POTS voice entity to a voice subscriber line rather than an FXS or FXO voice subscriber
line.
Syntax
send-ring
undo send-ring
View
POTS voice entity view, VoIP voice entity view, VoFR entity view
Default level
2: System level
40
Parameters
None
Examples
# Enable the local end to play ringback tone.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] send-ring
shutdown (voice entity view) Description
Use shutdown to change the management status of the specified voice entity from UP to DOWN.
Use undo shutdown to restore the default management status of the voice entity.
By default, the voice entity management status is UP.
Running shutdown will cause the voice entity unable to make calls.
Syntax
shutdown
undo shutdown
View
POTS voice entity view, VoIP voice entity view, VoFR entity view, IVR entity view
Default level
2: System level
Parameters
None
Examples
# Change the management status of voice entity 4 to DOWN.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 4 pots
[Sysname-voice-dial-entity4] shutdown
vad-on Description
Use vad-on to enable VAD.
Use undo vad-on to disable VAD.
By default, VAD is disabled.
If you execute vad-on or undo vad-on without specifying a codec, VAD for all codecs is enabled or disabled.
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.
g723r63: Specifies the g723r63 codec.
g729a: Specifies the g729a codec.
g729r8: Specifies the g729r8 codec.
Examples
# Enable VAD on POTS voice entity 10.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] vad-on
voice-setup Description
Use voice-setup to enter voice view and enable voice services.
Use undo voice-setup to disable voice services and quite voice view.
Syntax
voice-setup
undo voice-setup
View
System view
Default level
2: System level
42
Parameters
None
Examples
# Enter voice view and enable voice services.
<Sysname> system-view
[Sysname] voice-setup
voip timer Description
Use voip timer to set the time duration for switching from the current VoIP link to another VoIP link or a PSTN
link in case of a VoIP call failure.
Use undo voip timer to restore the default.
By default, the duration is five seconds.
For more information about call backup, see Voice Configuration Guide.
Syntax
voip timer voip-to-pots time
undo voip timer voip-to-pots
View
Voice view
Default level
2: System level
Parameters
voip-to-pots time: Specifies the time duration in seconds for switching from the current VoIP link to another
VoIP link or a PSTN link (that is, the call backup switching time) in case of a VoIP call failure, in the range of
3 to 30.
Examples
# Set the time duration for switching from the current VoIP link to another VoIP link or a PSTN link in case of
a VoIP call failure to 3 seconds.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] voip timer voip-to-pots 3
vqa dscp Description
Use vqa dscp to globally set the DSCP subfield in the ToS field in IP packets that carry the RTP stream or voice
signaling.
Use undo vqa dscp to restore the default.
By default, the DSCP subfield is set to ef, 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> system-view
[Sysname] voice-setup
[Sysname-voice] vqa dscp signal af41
vqa dsp-monitor buffer-time Description
Use vqa dsp-monitor buffer-time to set duration of monitoring DSP buffered data.
Use undo vqa dsp-monitor buffer-time to restore the default.
By default, the duration of monitoring DSP buffered data is 270 milliseconds.
Duration greater than 240 milliseconds is recommended because too small a duration value will result in
poor voice quality in the case of severe jitter.
Syntax
vqa dsp-monitor buffer-time time
undo vqa dsp-monitor buffer-time
View
Voice view
Default level
2: System level
Parameters
buffer-time time: Specifies the duration in milliseconds of monitoring DSP buffered data. The value is 0 or
ranges from 180 to 480.
Examples
# Set the duration of monitoring DSP buffered data to 300 milliseconds.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] vqa dsp-monitor buffer-time 300
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> system-view
[Sysname] voice-setup
[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> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] busytone-hookon timer 5
busytone-t-th Description
Use busytone-t-th to configure the number of busy tone periods for detection.
Use undo busytone-t-th to restore the default.
By default, the number of busy tone periods for detection is 2.
Enabling the busy tone detection is optional. Under particular situations, however, the actual busy tone data
cannot exactly match the busy tone parameters configured for the system. If there is a big difference, the busy
tone may not be detected correctly, resulting in on-hook failures or wrong on-hooks. By adjusting the time
threshold of busy tone detection, you make the busy tone detection more precise.
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
Analog FXO voice subscriber line view
Default level
2: System level
Parameters
time-threshold: Number of busy tone periods for detection, in the range of 2 to 12. A bigger value means a
longer busy tone detection time.
Examples
# Set the number of busy tone periods to 3.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] busytone-t-th 3
calling-name Description
Use calling-name to configure the calling name.
Use undo calling-name to remove the calling name.
By default, no calling name is configured.
Syntax
calling-name text
undo calling-name
View
Analog FXS voice subscriber line view
Default level
2: System level
Parameters
text: Name of the calling party associated with the FXS voice subscriber line, a string of 1 to 50 case-sensitive
characters including numbers 0 through 9, letters A through Z or a through z, underlines (_), hyphens (-),dots
(.), exclamation point (!), percent sign (%), asterisk (*), plus sign (+), grave accent (`), single quotation mark
(‗), and tilde (~).
Examples
# Configure the calling name on the FXS voice subscriber line 1/0 as tony.
<Sysname> system-view
[Sysname] subscriber-line 1/0
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
Analog FXS voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable CID on voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] cid display
cid receive Description
Use cid receive to enable CID.
Use undo cid receive to disable CID.
By default, CID is enabled.
When CID is disabled and the calling party sends a calling number, the local FXO interface performs these
actions:
If a number is configured in the number template for the POTS entity associated with the local FXO
interface, the interface substitutes this number for the calling number and sends it to the called side.
If wildcard dots (.) are used in the number configured in the number template for the POTS entity
associated with the local FXO interface, the interface substitutes zeros for the calling 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
Analog FXO voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable CID on voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] cid receive
cid ring Description
Use cid ring to configure the time for CID check and after the CID check, the number of rings the FXO line
receives before going off-hook.
Use undo cid ring to restore the default.
By default, CID check is performed between the first and the second rings, and the FXO line goes off-hook
as soon as the check completes, that is, cid ring 1 0.
Syntax
cid ring { 0 | 1 | 2 } [ times ]
undo cid ring
View
Analog FXO voice subscriber line view
Default level
2: System level
Parameters
0: CID check is performed before the phone rings.
1: CID check is performed between the first and the second rings.
2: CID check is performed between the second and the third rings.
times: Ring count after the CID check before the FXO line goes off-hook. The value is in the range 0 to 5. The
greater the value, the later the FXO line goes off-hook.
Examples
# Configure CID check to be performed before the phone rings on voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] cid ring 0
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> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] undo cid send
cid type Description
Use cid type to configure the format of message (which carries the calling number information) transmitted
over the FXS voice subscriber line.
Use undo cid type to restore the default message format.
By default, MDMF is adopted.
Two formats are available: MDMF and SDMF. If the remote end supports one format only, you must use the
same message format at the local end.
The calling name in the calling identity information can only be transmitted in MDMF format.
Syntax
cid type { complex | simple }
undo cid type
51
View
Analog FXS voice subscriber line view
Default level
2: System level
Parameters
complex: Calling identity information is transmitted in MDMF.
simple: Calling identity information is transmitted in SDMF.
Examples
# Set the format of the transmitted calling identity information to SDMF on voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] cid type simple
cng-on Description
Use cng-on to enable comfortable noise function.
Use undo cng-on to disable this function.
By default, the comfortable noise function is enabled.
You can use this command to generate a comfortable background noise to replace the toneless intervals
during a conversation.
Related commands: line and vad-on.
Syntax
cng-on
undo cng-on
View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice
subscriber line view
Default level
2: System level
Parameters
None
Examples
# Disable comfortable noise function on subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] undo cng-on
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
Code Country name (including customization)
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
Code Country name (including customization)
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> system-view
[sysname] voice-setup
[sysname-voice] cptone tone-type busy-tone amplitude 1200
default Description
Use default to restore the default settings for a voice subscriber line.
Syntax
default
View
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Restore the default settings for voice subscriber line 5/0.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] default
This command will restore the default settings. Continue? [Y/N]:y
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> system-view
[Sysname] voice-setup
[Sysname-voice] default subscriber-line receive gain 9.0
delay hold Description
Use delay hold to configure the delay signal duration in the delay start mode.
Use undo delay hold to restore the default.
By default, the delay signal duration is 400 milliseconds.
Related commands: em-signal.
Syntax
delay hold milliseconds
undo delay hold
View
E&M voice subscriber line view
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> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] em-signal delay
[Sysname-subscriber-line5/0] delay hold 500
delay rising Description
Use delay rising to configure a delay time from when the terminating side detects a seizure signal to when
it sends a delay signal in the delay start mode.
Use undo delay rising to restore the default.
By default, the delay time is 300 milliseconds.
Related commands: em-signal.
Syntax
delay rising milliseconds
undo delay rising
View
E&M voice subscriber line view
Default level
2: System level
Parameters
rising milliseconds: Specifies delay time (in milliseconds) from when the terminating side detects a seizure
signal to when it sends a delay signal in the delay start mode. The value ranges from 20 to 2,000.
Examples
# Set the delay time from when the terminating side detects a seizure signal to when it sends a delay signal
in the delay start mode to 700 milliseconds.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] em-signal delay
[Sysname-subscriber-line5/0] delay rising 700
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.
Related commands: em-signal.
Syntax
delay send-dtmf milliseconds
undo delay send-dtmf
View
E&M voice subscriber line view
Parameters
send-dtmf milliseconds: Specifies a delay (in milliseconds) before the originating side sends DTMF signals in
the immediate start mode. The value ranges from 50 to 5,000.
Examples
# Set the delay before the originating side sends DTMF signals in the immediate start mode to 3,000
milliseconds.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] delay send-dtmf 3000
delay send-wink Description
Use delay send-wink to configure an interval from when the terminating side receives a seizure signal to
when it sends a wink signal in the wink start mode.
Use undo delay send-wink to restore the default.
By default, the interval from when the terminating side receives a seizure signal to when it sends a wink
signal is 200 milliseconds in the wink start mode.
Related commands: em-signal.
Syntax
delay send-wink milliseconds
undo delay send-wink
View
E&M voice subscriber line view
Default level
2: System level
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> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] em-signal wink
[Sysname-subscriber-line5/0] delay send-wink 700
delay wink-hold Description
Use delay wink-hold to configure duration the terminating side sends wink signals in the wink start mode.
Use undo delay wink-hold to restore the default.
By default, the duration the terminating side sends wink signals is 500 milliseconds in the wink start mode.
Related commands: em-signal.
Syntax
delay wink-hold milliseconds
undo delay wink-hold
View
E&M voice subscriber line view
Default level
2: System level
Parameters
wink-hold milliseconds: Specifies duration (in milliseconds) the terminating side sends wink signals in the
wink start mode. The value ranges from 100 to 3,000.
Examples
# Set the duration the terminating side sends wink signals in the wink start mode to 700 milliseconds.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] em-signal wink
[Sysname-subscriber-line5/0] delay wink-hold 700
delay wink-rising Description
Use delay wink-rising to configure a maximum amount of time the originating side waits for a wink signal
after sending a seizure signal in the wink start mode.
Use undo delay wink-rising to restore the default.
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.
Related commands: em-signal.
Syntax
delay wink-rising milliseconds
undo delay wink-rising
View
E&M voice subscriber line view
Default level
2: System level
Parameters
wink-rising milliseconds: Specifies the maximum amount of time (in milliseconds) the originating side waits
for a wink signal after sending a seizure signal in the wink start mode. The value ranges from 100 to 5,000.
Examples
# Set the maximum amount of time the originating side waits for a wink signal after sending a seizure signal
in the wink start mode to 2,000 milliseconds.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] em-signal wink
[Sysname-subscriber-line5/0] delay wink-rising 2000
delay start-dial Description
Use delay start-dial to configure the dial delay.
Use undo delay start-dial to restore the default.
By default, the dial delay is 1 second.
Syntax
delay start-dial seconds
undo delay start-dial
View
FXS voice subscriber line view, FXO voice subscriber line view
Default level
2: System level
Parameters
seconds: Dial delay in seconds, in the range of 0 to 10.
Examples
# Set the dial delay on FXS subscriber line 1/0 to 5 seconds.
<Sysname> system-view
[Sysname] subscriber-line 1/0
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
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice
subscriber line view
Default level
2: System level
Parameters
string: Description string of voice subscriber line, whose length ranges from 1 to 80 characters.
Examples
# Mark voice subscriber line 1/0 as lab_1.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] description lab_1
disconnect lcfo
Description
Use disconnect lcfo to enable the sending of pulse signals at hangup.
Use undo disconnect lcfo to disable the sending of pulse signals at hangup.
By default, the sending of pulse signal at hangup is disabled, and the system plays busy tones to the other
end.
Syntax
disconnect lcfo
undo disconnect lcfo
View
FXS voice subscriber line view
Default Level
2: System level
62
Parameters
None
Examples
# Enable the sending of pulse signals at hangup on the FXS voice subscriber line 5/1.
<Sysname> system-view
[Sysname] subscriber-line 5/1
[Sysname-subscriber-line5/1] disconnect lcfo
display voice subscriber-line Description
Use display voice subscriber-line to view the configuration information of the subscriber line, such as the type,
status, codec mode, receive and transmit gains.
Related commands: subscriber-line.
Syntax
display voice subscriber-line line-number [ | { begin | exclude | include } regular-expression ]
View
Any view
Default level
2: System level
Parameters
line-number: Subscriber line number.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
NOTE:
Actual output information may vary depending on the device model.
# Display the configuration information about E&M voice subscriber line 5/0.
<Sysname> display voice subscriber-line 5/0
Current information ----- 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
Table 23 Output description
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
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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> system-view
[Sysname] voice-setup
[Sysname-voice] dtmf amplitude -8.0
dtmf sensitivity-level Description
Use dtmf sensitivity-level to set the DTMF detection sensitivity level and the absolute frequency deviation
when the DTMF detection sensitivity level is set to medium.
Use undo dtmf sensitivity-level to restore the default detection sensitivity level.
By default, the DTMF detection sensitivity level is low.
The following table shows the command and router compatibility:
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> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] dtmf sensitivity-level high
dtmf time Description
Use dtmf time to configure the related time parameters of DTMF.
Use undo dtmf time to restore the default.
By default, the persisting time of sending DTMF and the interval for sending DTMF are both 120 milliseconds.
The configuration will apply to the whole interface once you carry out the command.
Syntax
dtmf time { interval | persist } milliseconds
undo dtmf time { interval | persist }
View
Voice view
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> system-view
[Sysname] voice-setup
[Sysname-voice] dtmf time persist 200
[Sysname-voice] dtmf time interval 300
dtmf threshold Description
Use dtmf threshold to configure the sensitivity of DTMF digit detection.
Use undo dtmf threshold to restore the default.
The dtmf threshold command issues the thresholds for DTMF dial tone detection to the underlying layer DSP,
for the purpose of tuning detection sensitivity and reliability of the device subtly. Inside the DSP, a set of
generic default values have been configured. They are 1,400, 458, -9, -9, -9, -9, -3, -12, -12, 30, 300, 3,200,
375, with their index being 0 through 12. Professionals can use this command to adjust the device when
DTMF digit detection fails. In normal cases, the defaults are adopted.
Syntax
dtmf threshold analog index value
undo dtmf threshold analog index
View
Analog FXS voice subscriber line view, analog FXO voice subscriber line view, analog E&M voice subscriber
line view
Default level
2: System level
Parameters
analog: Analog voice subscriber line.
index: Index number corresponding to a threshold, an integer 0 through 12.
value: Threshold corresponding to the specified index. The value range varies with indexes. For details, see
Table 24.
According to the energy level of the row and column frequencies as well as the energy level of their double
frequencies, the system determines whether the input DTMF digit is valid.
The maximum energy of the input signal in the row frequency group is ROWMAX and the corresponding
double frequency energy is ROW2nd. The maximum energy in the column frequency group is COLMAX and
the corresponding double frequency energy is COL2nd.
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,
the higher the detection
reliability is. However,
the sensitivity decreases.
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
The larger the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
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.
–18 to –3 dB, with a
default of –9 dB
The smaller the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
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.
–18 to –3 dB, with a
default of –9 dB
The smaller the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
5
Upper limit of the ratio of the second largest
energy level from the column frequency group to
COLMAX. The ratio must be lower than this limit
for the input signal to be recognized as a DTMF
digit.
–18 to –3 dB, with a
default of –9 dB
The smaller the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
6
Upper limit of ROW2nd/ROWMAX. An input
signal is recognized as a DTMF digit only when
ROW2nd/ROWMAX < 6.
–18 to –3 dB, with a
default of –3 dB
The smaller the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
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.
–18 to –3 dB, with a
default of –12 dB
The smaller the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
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.
–18 to –3 dB, with a
default of –12 dB
The smaller the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
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
The larger the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
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
The larger the value is,
the higher the detection
reliability is. However,
the sensitivity decreases.
Examples
# Set the DTMF threshold 9 for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] dtmf threshold analog 9 40
echo-canceller Description
Use echo-canceller to enable echo cancellation and set the echo duration.
Use undo echo-canceller to disable the EC function.
By default, the EC function is enabled.
Related commands: subscriber-line and echo-canceller parameter.
NOTE:
The echo-canceller tail-length command is applicable only after the echo-canceller enable command is
executed.
Syntax
echo-canceller { enable | tail-length milliseconds }
undo echo-canceller { enable | tail-length }
View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice
subscriber line view
Default level
2: System level
Parameters
enable: Enables the echo cancellation (EC) function.
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> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] echo-canceller enable
[Sysname-subscriber-line1/0] echo-canceller tail-length 24
echo-canceller parameter Description
Use echo-canceller parameter to configure echo cancellation parameters.
Use undo echo-canceller parameter to restore the default.
By default, the convergence rate of comfort noise amplitude is 0, the maximum amplitude of comfort noise
is 256, the comfort noise mixture proportion control factor is 100, and the threshold of two-way talk is 1.
Related commands: echo-canceller.
Syntax
echo-canceller parameter { convergence-rate value | max-amplitude value | mix-proportion-ratio value |
talk-threshold value }
undo echo-canceller parameter { convergence-rate | max-amplitude | mix-proportion-ratio |
talk-threshold }
View
Voice view
Parameters
convergence-rate value: Sets the convergence rate of comfort noise amplitude. It ranges from 0 to 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> system-view
[Sysname] voice-setup
[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> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] em-phy-parm 2-wire
em-signal Description
Use em-signal to configure a start mode for an analog E&M voice subscriber line.
Use undo em-signal to restore the default start mode.
By default, the immediate start mode is selected for the analog E&M subscriber line.
Syntax
em-signal { delay | immediate | wink }
undo em-signal
View
Analog E&M voice subscriber line view
Default level
2: System level
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> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] em-phy-parm 4-wire
em-passthrough Description
Use em-passthrough to enable E&M analog control signals pass-through.
Use undo em-passthrough to disable E&M analog control signals pass-through.
By default, E&M analog control signals pass-through is disabled.
Syntax
em-passthrough
undo em-passthrough
View
Analog E&M voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable E&M analog control signals pass-through for E&M voice subscriber line 6/0.
<Sysname> system-view
[Sysname] subscriber-line 6/0
[Sysname-subscriber-line6/0] em-passthrough
hookoff-mode Description
Use hookoff-mode to configure the off-hook mode for the FXO voice subscriber line.
Use undo hookoff-mode to restore the default.
By default, the FXO voice subscriber line operates in the immediate off-hook mode.
73
Syntax
hookoff-mode { delay | immediate }
undo hookoff-mode
View
Analog FXO voice subscriber line view
Default level
2: System level
Parameters
delay: Specifies the FXO voice subscriber line to operate in the delay off-hook mode.
immediate: Specifies the FXO voice subscriber line to operate in the immediate off-hook mode.
Examples
# Specify an FXO voice subscriber line to operate in the delay off-hook mode.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line 1/0] hookoff-mode delay
hookoff-mode delay bind Description
Use hookoff-mode delay bind to bind an FXS voice subscriber line to the FXO voice subscriber line.
Use undo hookoff-mode delay bind to remove the binding.
By default, no FXS voice subscriber line is bound to the FXO voice subscriber line.
After an FXS voice subscriber line is bound to the FXO voice subscriber line, the off-hook/on-hook state of
these two lines will be consistent.
NOTE:
To keep the consistent off-hook/on-hook state between the bound FXS and FXO voice subscriber lines, you must
consider the configurations of the private-line and caller-permit commands when executing the hookoff-mode
delay bind fxs_subscriber_line command. The FXS voice subscriber line specified by fxs_subscriber_line must be
the one to which the dedicated line number points. In addition, only the bound FXS voice subscriber line is allowed
to originate calls to the FXO voice subscriber line by restricting incoming calls. For more information about the
private-line and caller-permit command, see the chapter “Dial plan configuration commands.”
The bound FXS and FXO voice subscriber lines must come from the same device.
Use ring-immediately keyword to quicken ringing synchronization between the FXO voice subscriber line and its
bound FXS voice subscriber line. However, for the telephone supporting calling identification display, the calling
number will be displayed after the second ringing tone.
Syntax
hookoff-mode delay bind fxs_subscriber_line [ ring-immediately ]
undo hookoff-mode delay bind
View
Analog FXO voice subscriber line view
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> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] hookoff-mode delay bind 3/0
hookoff-time Description
Use hookoff-time to configure the on-hook timer length.
Use undo hookoff-time to restore the default on-hook timer length.
By default, no on-hook timer length is set.
Syntax
hookoff-time time
undo hookoff-time
View
Analog FXO voice subscriber line view
Default level
2: System level
Parameters
time: Length of the on-hook timer in seconds, in the range of 60 to 36,000.
Examples
# Set the on-hook timer length to 500 seconds.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] hookoff-time 500
impedance Description
Use impedance to set the current electrical impedance on an FXO or FXS voice subscriber line.
Use undo impedance to restore the default.
By default, the electrical impedance on the FXO or FXS voice subscriber line is the impedance value
corresponding to China.
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.
r600: 600-ohm real impedance.
r650: 650-ohm real impedance.
r700: 700-ohm real impedance.
r750: 750-ohm real impedance.
r800: 800-ohm real impedance.
r850: 850-ohm real impedance.
r900: 900-ohm real impedance.
r950: 950-ohm real impedance.
Examples
# Set the current electric impedance to r600 on voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] impedance r600
nlp-on Description
Use nlp-on to enable the EC nonlinear processing function on a voice interface.
Use undo nlp-on to disable the function.
By default, the EC nonlinear processing function is enabled.
The following table shows the command and router compatibility:
76
Command A-MSR900 A-MSR20-1X A-MSR20 A-MSR30 A-MSR50
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
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Disable the EC nonlinear processing function on voice interface.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line 1/0] undo nlp-on
open-trunk Description
Use open-trunk to enable E&M non-signaling mode.
Use undo open-trunk to disable the E&M non-signaling mode.
By default, the E&M non-signaling mode is disabled.
77
Syntax
open-trunk { caller monitor interval | called }
undo open-trunk
View
E&M voice subscriber line view
Default level
2: System level
Parameters
caller monitor interval: Enables the local voice gateway to use E&M non-signaling mode when serving as the
calling side and specifies the monitoring interval in the range 60 to 600 seconds.
called: Enables the local voice gateway to use E&M non-signaling mode when serving as the called side.
Examples
# Configure the PLAR function on voice subscriber line 5/0 so that 100 is automatically dialed out when the
subscriber picks up the phone. Enable E&M non-signaling mode on the calling voice gateway and specify
the monitoring interval as 120 seconds.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] private-line 100
[Sysname-subscriber-line5/0] open-trunk caller monitor 120
plc-mode Description
Use plc-mode to configure a packet loss compensation mode for the analog FXS/FXO voice subscriber line.
Use undo plc-mode to restore the default.
By default, the gateway-specific algorithm is used for packet loss compensation.
Syntax
plc-mode { general | specific }
undo plc-mode
View
Analog FXS voice subscriber line view, analog FXO voice subscriber line view
Default level
2: System level
Parameters
general: Uses the universal frame erasure algorithm.
specific: Uses the specific algorithm provided by the voice gateway.
Examples
# Configure the voice gateway to use the universal packet loss compensation algorithm.
<Sysname> system-view
[Sysname] subscriber-line 1/0
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
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice
subscriber line view
Default level
2: System level
Parameters
value: Voice input gain in dB, in the range of -14.0 to +13.9 with one digit after the decimal point.
Examples
# Set the voice input gain to 3.5 dB on subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] receive gain 3.5
reset voice cmc statistic Description
Use reset voice cmc statistic to clear calling statistics on the CMC module.
Related commands: display voice cmc.
Syntax
reset voice cmc statistic
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.
The following table shows the command and router compatibility:
Command A-MSR900 A-MSR20-1X A-MSR20 A-MSR30 A-MSR50
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
Analog FXO voice subscriber line view
Default level
2: System level
Parameters
value: Debounce time of ring detection, in milliseconds, in the range of 4 to 15.
Examples
# Configure the debounce time of ring detection on FXO voice subscriber line 1/0 to 15 milliseconds.
<sysname> system-view
[sysname] subscriber-line 1/0
81
[sysname-subscriber-line1/0] ring-detect debounce 15
ring-detect frequency Description
Use ring-detect frequency to set the frequency value in ring detection.
Use undo ring-detect frequency to restore the default.
By default, the frequency in the ring detection is 40 Hz.
The following table shows the command and router compatibility:
Command A-MSR900 A-MSR20-1X A-MSR20 A-MSR30 A-MSR50
ring-detect frequency The following voice modules support this command:
SIC-2FXS1FXO
Syntax
ring-detect frequency value
undo ring-detect frequency
View
Analog FXO voice subscriber line view
Default level
2: System level
Parameters
value: Frequency value in the ring detection, in Hz. The value is in the range 30 to 100 with the step of 10.
Examples
# Set the frequency value in ring detection on FXO voice subscriber line 1/0 to 100 Hz.
<sysname> system-view
[sysname] subscriber-line 1/0
[sysname-subscriber-line1/0] ring-detect frequency 100
send-busytone Description
Use send-busytone to enable busy tone sending on the FXO interface. Use undo send-busytone to disable
busy tone sending on the FXO interface.
By default, busy tone sending is disabled.
Syntax
send-busytone { enable | time seconds }
undo send-busytone { enable | time }
View
Analog FXO voice subscriber line view
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> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] send-busytone enable
[Sysname-subscriber-line1/0] send-busytone time 5
shutdown (voice subscriber line view) Description
Use shutdown to set the voice subscriber line DOWN.
Use undo shutdown to restore the default status of the voice subscriber line.
By default, the voice subscriber line is UP.
The POTS interface on the voice interface card will be DOWN and there will be no sound on the connected
telephone after shutdown is executed, and whereas the specified voice subscriber line will be UP after undo
shutdown is executed.
Syntax
shutdown
undo shutdown
View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice
subscriber line view
Default level
2: System level
Parameters
None
Examples
# Shut down voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] shutdown
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silence-th-span Description
Use silence-th-span to set the silence duration for automatic on-hook.
Use undo silence-th-span to restore the default.
By default, the silence threshold is 20 and the silence duration for automatic on-hook is 7,200 seconds (2
hours).
Syntax
silence-th-span threshold time-length
undo silence-th-span
View
Analog FXO subscriber line view
Default level
2: System level
Parameters
threshold: Silence threshold. If the amplitude of voice signals from the switch is smaller than this value, the
system regards the voice signals as silence. This threshold ranges from 0 to 200. Normally, the signal
amplitude on the links without traffic is in the range of 2 to 5.
time-length: Silence duration for automatic on-hook. Upon expiration of this duration, the system performs
on-hook automatically. It ranges from 2 to 7,200 seconds.
Examples
# Set the silence threshold to 20 and the silence duration to 10 seconds.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] silence-th-span 20 10
slic-gain Description
Use slic-gain to configure the output gain of the SLIC chip. The bottom layer tunes the signal gain through the
SLIC chip.
Use undo slic-gain to restore the default output gain.
By default, the output gain of the SLIC chip is 0 dB.
Syntax
slic-gain { 0 | 1 }
undo slic-gain
View
Analog E&M voice subscriber line view
Default level
2: System level
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Parameters
0: Sets the output gain of the SLIC chip to 0.8 dB.
1: Sets the output gain of the SLIC chip to 2.1 dB.
Examples
# Set SLIC-gain to 1 in analog E&M voice subscriber line view.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] slic-gain 1
subscriber-line Description
Use subscriber-line to enter the specified voice subscriber line view.
Use subscriber-line line-number to enter the voice subscriber line view. For example, if line-number is an FXS
voice subscriber line, the system will enter the FXS voice subscriber line view; if line-number is an analog
E&M voice subscriber line, the system will enter analog E&M voice subscriber line view.
Syntax
subscriber-line line-number
View
System view
Default level
2: System level
Parameters
line-number: Voice subscriber line number.
Examples
# Enter the view of the voice subscriber line 1/0 in system view.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0]
timer dial-interval Description
Use timer dial-interval to configure the maximum interval for dialing the next digit.
Use undo timer dial-interval to restore the default setting.
By default, the maximum interval for dialing the next digit is 10 seconds.
This timer will restart each time the subscriber dials a digit and will work in this way until all the digits of the
number are dialed. If the timer expires before the dialing is completed, the subscriber will be prompted to
hook up and the call is terminated.
Syntax
timer dial-interval seconds
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> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] timer dial-interval 5
timer disconnect-pulse
Description
Use timer disconnect-pulse to configure the time duration for the sending of the pulse signals at hangup.
Use undo timer disconnect-pulse to restore the default setting.
Syntax
timer disconnect-pulse milliseconds
undo timer disconnect-pulse
View
FXS voice subscriber line view
Default Level
2: System level
Parameters
milliseconds: Time duration (in milliseconds) for the sending of the pulse signals at hangup. The value is in
the range 0 to 1500, and defaults to 750.
Examples
# Configure the time duration for the sending of the pulse signals at hangup on the FXS voice subscriber line
5/1 as 1 second.
<Sysname> system-view
[Sysname] subscriber-line 5/1
[Sysname-subscriber-line5/1] timer disconnect-pulse 1000
timer first-dial Description
Use timer first-dial to configure the maximum interval between off-hook and dialing the first digit.
Use undo timer first-dial to restore the default setting.
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By default, the maximum interval between off-hook and dialing the first digit is 15 seconds.
Upon the expiration of the timer, the subscriber will be prompted to hook up and the call is terminated.
Syntax
timer first-dial seconds
undo timer first-dial
View
FXS voice subscriber line view, FXO voice subscriber line view
Default level
2: System level
Parameters
seconds: Maximum interval in seconds between off-hook and dialing the first digit, in the range of 1 to 300.
Examples
# Set the maximum interval between off-hook and dialing the first digit to 10 seconds.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] timer first-dial 15
timer hookflash-detect Description
Use timer hookflash-detect to configure the time range for the duration of an on-hook condition that will be
detected as a hookflash.
Use undo timer hookflash-detect to restore the default.
By default, the time range is 50 to 180 milliseconds, that is, if an on-hook condition that lasts for a period that
falls within the hookflash duration range is considered a hookflash.
Syntax
timer hookflash-detect hookflash-range
undo timer hookflash-detect
View
Analog FXS subscriber line view
Default level
2: System level
Parameters
hookflash-range: Hookflash duration range, in milliseconds, in the range of 50 to 1,200.
Examples
# Set the hookflash duration range for voice subscriber line 1/0 to 100 to 200 milliseconds.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] timer hookflash-detect 100-200
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
Analog FXO voice subscriber line view
Default level
2: System level
Parameters
milliseconds: Interval between on-hook and off-hook in milliseconds, in the range of 500 to 4,000.
Examples
# Set the interval from on-hook to off-hook for FXO voice subscriber line 1/0 to 600 milliseconds.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] timer hookoff-interval 600
timer ring-back Description
Use timer ring-back to configure the maximum duration of playing the ringback tone.
Use undo timer ring-back to restore the default.
By default, the maximum duration of playing the ringback tone is 60 seconds.
Syntax
timer ring-back seconds
undo timer ring-back
View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view
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> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] timer ring-back 8
timer wait-digit Description
Use timer wait-digit to configure the maximum time duration the system waits for a digit.
Use undo timer wait-digit to restore the default time settings.
By default, the maximum time duration the system waits for a digit is 5 seconds.
Syntax
timer wait-digit { seconds | infinity }
undo timer wait-digit
View
E&M voice subscriber line view
Default level
2: System level
Parameters
seconds: Maximum duration in seconds the system waits for a digit, in the range of 3 to 600.
infinity: Infinite time.
Examples
# Set the maximum duration waiting for the first dial on voice line 5/0 to 5 seconds.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] timer wait-digit 5
transmit gain Description
CAUTION:
Gain adjustment may lead to call failures. You are not recommended to adjust the gain. If necessary, do
it under the guidance of technical personnel.
Use transmit gain to set the voice subscriber line output end gain value.
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Use undo transmit gain to restore the default value.
By default, the output gain on the voice interface is 0 dB.
This command is applicable to FXO, FXS, E&M, BSV and E1/T1 voice subscriber lines.
When a relatively small voice signal power is needed on the output line, this command can be used to
properly increase the voice output attenuation value.
Related commands: receive gain and subscriber-line.
Syntax
transmit gain value
undo transmit gain
View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice
subscriber line view
Default level
2: System level
Parameters
value: Voice output gain in dB, in the range of -14.0 to 13.9 with one digit after the decimal point.
Examples
# Set the voice output gain value to –6.7dB on subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] transmit gain -6.7
type Description
Use type to configure the analog E&M subscriber line signal type.
Use undo type to cancel the existing settings.
By default, the analog E&M subscriber line signal type is type 5.
This command is only applicable to an E&M subscriber line, and once configured, is effective on all analog
E&M lines in the corresponding slot.
Syntax
type { 1 | 2 | 3 | 5 }
undo type
View
Analog E&M voice subscriber line view
Default level
2: System level
Parameters
1, 2, 3 and 5: Correspond respectively to the four signal types of analog E&M subscriber lines.
90
Examples
# Configure subscriber line 5/0 analog E&M subscriber line type as type 3.
<Sysname> system-view
[Sysname] subscriber-line 5/0
[Sysname-subscriber-line5/0] type 3
vi-card busy-tone-detect Description
Use vi-card busy-tone-detect to configure the parameters for the busy tone detection on the FXO interface.
Use undo vi-card busy-tone-detect to restore the default settings.
This command applies to the FXO interface only.
The system supports four types of busy tones, which are specified by the index argument.
When detecting a busy tone on the FXO interface, the system will automatically calculate the parameters
related to busy tone detection. You can use the display current-configuration command to display the
settings of these parameters.
After you use the vi-card busy-tone-detect custom command to configure the parameters related to the busy
tone detection, these parameters do not take effect immediately. The manually configured busy tone
parameters can take effect only after you execute the area custom command in voice view.
Syntax
vi-card busy-tone-detect { auto index line-number | custom area-number index argu f1 f2 p1 p2 p3 p4 p5
p6 p7 }
undo vi-card busy-tone-detect { auto | custom } index
View
Voice view
Default level
2: System level
Parameter
Index: Index of busy tone type, in the range of 0 to 3.
line-number: Voice subscriber line number. The value range varies with devices as well as the cards inserted.
area-number: Area number. It is set to 2.
argu: Reserved, in the range of 0 to 32,767.
f1: Frequency 1 in Hz, in the range of 50 to 3,600.
f2: Frequency 2 in Hz, in the range of 50 to 3,600.
p1: Signal amplitude 1, in the range of 50 to 32,767.
p2: Signal amplitude 2, in the range of 50 to 32,767.
p3: Duration of a single tone in milliseconds, in the range of 10 to 1,000.
p4: Duration error of a single tone in milliseconds, in the range of 0 to 500.
p5: Duration of silence in milliseconds, in the range of 10 to 1,000.
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p6: Duration error of silence in milliseconds, in the range of 0 to 500.
p7: Absolute difference between p3 and p5 in milliseconds, in the range of 0 to 500
Examples
# Enable the automatic busy tone detection on subscriber line 2/0, with the busy tone index being 0.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] vi-card busy-tone-detect auto 0 2/0
# Manually configure busy tone indexed as 0, duration limit of high/low level, duration error of high/low
level, and duration difference of high/low level.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] vi-card busy-tone-detect custom 2 1 99 450 450 8000 8000 800 300 500 500 500
vi-card cptone-custom Description
Use vi-card cptone-custom to configure parameters for a customized call progress tone.
Use undo vi-card cptone-custom to remove the configuration.
By default, no customized call progress tone is configured.
After you configure parameters for a customized call progress tone, they do not take effect immediately. They
do only after you execute cptone country-type CS in voice view.
Syntax
vi-card cptone-custom { busy-tone | congestion-tone | dial-tone | ringback-tone | special-dial-tone |
waiting-tone } comb freq1 freq2 time1 time2 time3 time4
undo vi-card cptone-custom { all | busy-tone | congestion-tone | dial-tone | ringback-tone |
special-dial-tone | waiting-tone }
View
Voice view
Default level
2: System level
Parameters
busy-tone: Busy tone.
congestion-tone: Congestion tone.
dial-tone: Dial tone.
ringback-tone: Ringback tone.
special-dial-tone: Special dial tone.
waiting-tone: Call waiting tone.
comb: Combination mode, in the range of 0 to 2. The values 0, 1, and 2 represent the superimposition and
modulation of two frequencies, and alternation between two frequencies, respectively.
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freq1 and freq2: Two frequencies in Hz. The frequency range is related to the combination mode. In the case
of frequency superimposition or alternation, the two frequencies fall in the range of 300 Hz to 3,400 Hz. In
the case of frequency modulation, the two frequencies fall in the range of 300 Hz to 3,400 Hz, and the sum
of and the absolute difference between the two frequencies also fall in this range.
time1: Make time for the first make-to-break ratio in milliseconds, in the range of 30 to 8,192. In the case of
continuous play, the value is 8,192.
time2: Break time for the first make-to-break ratio in milliseconds, 30 through 8,191.
time3: Make time for the second make-to-break ratio in milliseconds, 30 through 8,191.
time4: Break time for the second make-to-break ratio in milliseconds, 30 to 8,191.
Example
# Customize parameters for a busy tone, with the two frequencies both being 425 Hz, and the make time
and break time both being 350 milliseconds.
<sysname> system-view
[sysname] voice-setup
[sysname-voice] vi-card cptone-custom busy-tone 0 425 425 350 350 350 350
vi-card reboot Description
Use vi-card reboot to reboot a voice card.
First use display version or display device to display the distributed slots of the voice cards in the router.
Related commands: display version and display device (Fundamentals Command Reference).
NOTE:
The vi-card reboot command can be used to reboot all analog voice cards (including FXS, FXO, and E&M),
SIC-AUDIO, and BSV.
The SIC digital voice cards and VE1 and VT1 voice cards cannot be rebooted by using commands.
You can use the reboot slot slot-number command to reset the analog voice card of FIC. For more information about
the reboot slot command, see Fundamentals Command Reference.
Syntax
vi-card reboot slot-number
View
Voice view
Default level
2: System level
Parameters
slot-number: Number of the slot where the voice card is located.
Examples
# Reset the voice card of slot 3.
<Sysname> system-view
[Sysname] voice-setup
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> system-view
[Sysname] subscriber-line 1/0:1
[Sysname-subscriber-line1/0:1] amd enable
amd parameter Description
Use amd parameter to configure AMD parameters.
Use undo amd parameter to restore the default.
By default, the machine-time keyword is 2600 milliseconds; the max-analyze-time keyword is 4000
milliseconds; the min-silence-time keyword is 800 milliseconds; the valid-voice-time keyword is 120
milliseconds; the voice-energy-threshold keyword is 100.
There are four AMD detection results: voice, automatic, silence and unknown.
Syntax
amd parameter { machine-time value | max-analyze-time value | min-silence-time value | valid-voice-time
value | voice-energy-threshold value }
undo amd parameter { machine-time | max-analyze-time | min-silence-time | valid-voice-time |
voice-energy-threshold }
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> system-view
[Sysname] voice-setup
[Sysname-voice] amd parameter max-analyze-time 5000
ani Description
Use ani to enable the terminating point to request calling party information (service category and calling
number) from the originating point during call connection.
Use undo ani to disable the terminating point from requesting calling party information from the originating
point.
By default, the terminating point does not request calling party information from the originating point during
call connection.
Related commands: cas and ani-offset.
NOTE:
Configure the local end with this command to support automatic number identification.
This command applies to R2 signaling only.
Normally the all keyword is configured. Use ka keyword only when required by the connected switch to prevent call
failures.
Syntax
ani { all | ka }
undo ani
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] ani all
ani-offset Description
Use ani-offset to configure the number of called number digits that need to be collected prior to requesting
calling party information.
Use undo ani-offset to restore the default value.
Before adequate digits are collected, the system will wait for the next digit until the timer expires. During this
period, the system does not request calling party information. It does that only after adequate digits are
collected.
By default, the number of digits to be collected before receiving calling party information is 1.
This command applies to R2 signaling only.
Related commands: cas, timer, reverse, and renew.
NOTE:
Before you can configure this command, you must configure the ani command.
Syntax
ani-offset number
undo ani-offset
View
R2 CAS view
Default level
2: System level
Parameters
number: Number of digits to be collected, in the range of 1 to 10.
96
Examples
# Start requesting calling number or caller identifier after receiving three digits.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] ani all
[Sysname-cas1/0:0] ani-offset 3
answer enable Description
Use answer enable to configure the originating point to require the terminating point to send answer signal.
The two parties begin to talk only after the originating point receives an answer signal.
Use undo answer enable to restore the default.
By default, the originating party requires the terminating party to send answer signal.
This command applies to R2 signaling only.
The R2 line signaling coding schemes in some countries do not include answer signal sending. To
accommodate to such schemes, you must configure answer enable on the originating point. This allows the
terminating point to set up calls after a specified time period.
Related commands: re-answer enable and timer dl re-answer.
Syntax
answer enable
undo answer enable
View
R2 CAS view
Default level
2: System level
Parameters
None
Examples
# Configure the originating point to disable the terminating point from sending answer signals.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] undo answer enable
callmode Description
Use callmode to configure the connection mode for an R2 call.
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas 1/0:0] callmode segment
cas Description
Use cas to enter R2 CAS view, digital E&M signaling view, or digital LGS signaling view.
After entering a signaling view, you may configure signaling parameters as desired. When doing that,
assign the same value to the ts-set-number keyword in commands cas and timeslot-set.
Related commands: timeslot-set, ani-offset, reverse, select-mode, timer, trunk-direction, and renew.
Syntax
cas ts-set-number
View
E1 interface view, T1 interface view
Default level
2: System level
Parameters
ts-set-number: Number of a created timeslot (TS) group, in the range of 0 to 30. The number of a T1 timeslot
group ranges from 0 to 23.
Examples
# Enter the R2 CAS view of TS set 5.
<Sysname> system-view
98
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 5
clear-forward-ack enable Description
Use clear-forward-ack enable to enable the terminating point to respond with a clear-back signal when the
originating point (the calling party) disconnects a call.
Use undo clear-forward-ack enable to disable the terminating point from responding with a clear-back
signal when the originating point (the calling party) disconnects a call.
By default, the terminating point does not send clear-back signals to acknowledge clear-forward signals.
This command applies to R2 signaling only.
In some countries, if the terminating point controls trunk circuit reset in the R2 signaling exchange process,
when the calling party disconnects a call and the originating point sends a clear-forward signal to the
terminating point, the terminating point sends a clear-back signal as an acknowledgement, and then sends
a release guard signal to indicate that the line of the terminating point is thoroughly released.
During R2 line signaling exchange, trunk circuit reset is sometimes controlled by the called party (terminating
point). The practice in some countries in this case is that after the terminating point receives a clear-forward
signal from the originating point, it sends back a clear-back signal as an acknowledgement and then a
release-guard signal to indicate that the line at the terminating point side is fully released.
Related commands: mode.
Syntax
clear-forward-ack enable
undo clear-forward-ack enable
View
R2 CAS view
Default level
2: System level
Parameters
None
Examples
# Enable the terminating point to acknowledge clear-forward signals with clear-back signals.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] clear-forward-ack enable
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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the configuration of voice subscriber line 5/0:0.
<Sysname> display voice subscriber-line 5/0:0
Current information ----- 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
Table 25 Output description
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.
This command applies to R2 signaling only.
You may need to use this command to accommodate to the ABCD bit pattern schemes used in different
countries.
Related commands: seizure-ack enable and answer enable.
Syntax
dl-bits { answer | blocking | clear-back | clear-forward | idle | seize | seizure-ack | release-guard }
{ received | transmit } ABCD
undo dl-bits { answer | blocking | clear-back | clear-forward | idle | seize | seizure-ack | release-guard }
{ received | transmit }
View
R2 CAS view
Default level
2: System level
Parameters
answer: Answer signal of R2 line signaling.
blocking: Blocking signal of R2 line signaling.
clear-back: Clear-back signal of R2 line signaling.
clear-forward: Clear-forward signal of R2 line signaling.
idle: Idle signal of R2 line signaling.
seize: Seizure signal of R2 line signaling.
seizure-ack: Seizure acknowledgement signal of R2 line signaling.
release-guard: Release guard signal of R2 line signaling.
received: Indicates that the signaling setting applies to received R2 line signals.
transmit: Indicates that the signaling setting applies to transmitted R2 line signals.
ABCD: ABCD bit pattern of R2 line signals, in the range of 0000 to 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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] dl-bits idle received 1101
[Sysname-cas1/0:0] dl-bits idle transmit 1011
dtmf enable Description
Use dtmf enable to set the way receiving and transmitting R2 signals to DTMF mode.
Use undo dtmf enable to restore the default.
By default, MFC mode is adopted.
This command applies to R2 signaling only.
Related commands: timer dtmf.
Syntax
dtmf enable
undo dtmf enable
View
R2 CAS view
Default level
2: System level
Parameters
None
Examples
# Adopt DTMF mode to receive and send R2 signals.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] dtmf enable
dtmf threshold digital Description
Use dtmf threshold digital to set the DTMF detection sensitivity.
Use undo dtmf threshold digital to restore the default DTMF detection sensitivity.
By default, the DTMF detection sensitivity level is 0, that is, insensitive.
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The more sensitive the DTMF detection is, the larger the tolerance of DTMF collection is. The possibility of
detecting error codes becomes relatively high while the possibility of missing detecting error codes becomes
low.
Syntax
dtmf threshold digital value
undo dtmf threshold digital
View
BSV voice subscriber line view
Default level
2: System level
Parameters
digital: Sets a digital voice subscriber line.
value: 0 or 1. 0 indicates that DTMF detection is insensitive while 1 indicates that DTMF detection is sensitive.
Examples
# Set the DTMF detection to be sensitive.
<Sysname> system-view
[Sysname] subscirber-line1/0:0
[Sysname-subscriber-line1/0:0] dtmf threshold digital 1
enable snmp trap updown Description
Use enable snmp trap updown to enable the interface to generate linkUp/linkDown traps upon link
changes.
Use undo enable snmp trap updown to disable the interface to generate linkUp/linkDown traps upon link
changes.
By default, the interface is enabled to generate linkUp/linkDown traps upon link changes.
Syntax
enable snmp trap updown
undo enable snmp trap updown
View
BSV BRI interface view
Default level
2: System level
Parameters
None
Examples
# Disable interface BSV BRI 2/0 to generate linkUp/linkDown traps.
<Sysname> system-view
[Sysname] interface bri 2/0
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.
This command applies to R2 signaling only.
You may configure final-callednum to accommodate to the R2 interregister signaling in some countries
where a number terminator can be sent to indicate that all digits of a called number have been sent.
Related commands: register-value digital-end.
Syntax
final-callednum enable
undo final-callednum enable
View
R2 CAS view
Default level
2: System level
Parameters
None
Examples
# Enable the originating point to send the number terminator signal.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] final-callednum enable
force-metering enable Syntax
force-metering enable
undo force-metering enable
View
R2 CAS view
Default level
2: System level
105
Parameters
None
Examples
# Enable R2 metering signal processing.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] force-metering enable
group-b enable Description
Use group-b enable to enable R2 signaling to use Group B signals to complete registers exchange.
Use undo group-b enable to disable R2 signaling from using Group B signals to complete registers
exchange.
By default, Group B signals are used to complete registers exchange.
This command applies to R2 signaling only.
You may need to configure the undo form of this command to accommodate to the R2 interregister signaling
in some countries where Group B signals is not supported or cannot be interpreted correctly.
Related commands: register-value req-switch-groupb.
Syntax
group-b enable
undo group-b enable
View
R2 CAS view
Default level
2: System level
Parameters
None
Examples
# Adopt Group B signals to complete registers exchange.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] group-b enable
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
POTS voice entity view
Default level
2: System level
Parameters
slot-number: Number of the E1/T1 interface corresponding to a subscriber line.
ts-set-number: Number of the TS set created on the E1/T1 interface.
15: Indicates that the POTS voice entity is to be associated with an E1 voice ISDN PRI interface.
23: Indicates that the POTS voice entity is to be associated with a T1 voice ISDN PRI interface.
Examples
# Associate a POTS entity with a TS set on an E1 interface.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] line 1/0:1
mode Description
Use mode to configure a national R2 signaling variant.
Use undo mode to restore the default.
By default, ITU-T R2 signaling applies.
This command applies to R2 signaling only.
The R2 signaling standards implemented in different countries and regions may vary. They are called ITU
variants. To accommodate to the R2 signaling in a country or region, you may use the mode command. The
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system can automatically select the appropriate subscriber line state, service category, metering signal, and
signal values of C and D bits.
At present, the device supports Brazil, Mexico, Argentina, India, New Zealand, Thailand, Bengal, South
Korea, Hongkong, Indonesia, and other ITU-T variants.
With the default-standard keyword configured, the system initializes the subscriber line status, service type,
metering signal and C and D signaling bits and other parameters depending on the default settings of
configured national R2 signaling variants.
If the custom keyword is configured, you can customize specific signaling exchange procedures and signal
values in R2 signaling to accommodate to countries.
Related commands: register-value and force-metering enable.
Syntax
mode zone-name [ default-standard ]
undo mode
View
R2 CAS view
Default level
2: System level
Parameters
zone-name: Country or region name. The argument can be one of the specified values:
argentina: Uses Argentinean R2 signaling standard.
australia: Uses Australian R2 signaling standard.
bengal: Uses Bengalee R2 signaling standard.
brazil: Uses Brazilian R2 signaling standard.
china: Uses Chinese R2 signaling standard.
custom: Uses customized R2 signaling standard.
hongkong: Uses Hongkong R2 signaling standard.
india: Uses Indian R2 signaling standard.
indonesia: Uses Indonesian R2 signaling standard.
itu-t: Uses ITU-T R2 signaling standard.
korea: Uses Korean R2 signaling standard.
malaysia: Uses Malaysian R2 signaling standard.
mexico: Uses Mexican R2 signaling standard.
newzealand: Uses New Zealand R2 signaling standard.
singapore: Uses Singaporean R2 signaling standard.
thailand: Uses Thai R2 signaling standard.
default-standard: Initializes R2 signaling parameters such as values of the force-metering command based
on national R2 signaling variants.
Examples
# Adopt Hongkong default R2 signaling.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] mode hongkong default-standard
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
Voice subscriber line view
Default level
2: System level
Parameters
a-law: Companding A-law, used in most part of the world other than North America and Japan, such as
China, Europe, Africa, and South America.
µ-law: Companding µ-law, used in North America and Japan.
Examples
# Adopt µ-law companding for signal quantization.
<Sysname> system-view
[Sysname] subscirber-line1/0:0
[Sysname-subscriber-line1/0:0] pcm u-law
posa called-length Description
Use posa called-length to set the length of called numbers that can be received by the E1POS card.
Use undo posa called-length to restore the default.
By default, the length of called numbers that can be received by the E1POS card is 31 digits.
Syntax
posa called-length called-length
undo posa called-length
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> system-view
[Sysname] controller e1 5/0
[Sysname-e1 5/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 5/0] cas 0
[Sysname-cas5/0:0] posa called-length 8
pri-set Description
Use pri-set to bundle timeslots on an E1 or T1 interface into a PRI group.
Use undo pri-set to remove the bundle.
By default, no PRI group is created.
When creating a PRI group on a CE1/PRI interface:
TS0 is used for FSC, TS16 as a D channel for signaling transmission, and other timeslots as B channels
for data transmission. You may bind the timeslots except for timeslot 0 into a PRI group (as the D
channel, timeslot 16 is automatically bundled). This PRI group is logically equivalent to an ISDN PRI
interface in the form of 30B + D. If no timeslot is specified, all timeslots except for TS0 are bound into
an interface similar to an ISDN PRI interface in the form of 30B+D.
For the created PRI group, the system automatically creates a serial interface named serial number:15.
When creating a PRI group on a T1 interface:
TS24 is used as D channel for signaling transmission, and other timeslots as B channels for data
transmission. You may randomly bind these timeslots into a PRI group (as the D channel, TS24 is
automatically bound). This PRI group is logically equivalent to an ISDN PRI interface in the form of 23B
+ D.
For the created PRI group, the system automatically creates a serial interface named serial number:23.
Syntax
pri-set [ timeslot-list range ]
undo pri-set
View
E1 interface view, T1 interface view
Default level
2: System level
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] pri-set timeslot-list 1,2,8-12
qsig-tunnel enable Description
Use qsig-tunnel enable to enable the QSIG tunneling function.
Use undo qsig-tunnel enable to disable the function.
By default, the QSIG tunneling function is disabled.
Syntax
qsig-tunnel enable
undo qsig-tunnel enable
View
Digital voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable the QSIG tunneling function.
<Sysname> system-view
[Sysname] subscriber-line 1/0:15
[Sysname-subscriber-line1/1:15] qsig-tunnel enable
re-answer enable Description
Use re-answer enable to enable the originating point to support re-answer signal processing.
Use undo re-answer enable to restore the default.
By default, the originating point does not support re-answer signal processing.
This command applies to R2 signaling only.
In some countries, re-answer process is needed in R2 signaling. When the terminating point sends a
clear-back signal, the originating point does not release the line right away, but maintains the call state
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas 1/0:0] re-answer enable
register-value Description
Use register-value to configure R2 register signal values.
Use undo register-value to restore the defaults.
You may set a signal value to 16 to indicate that the signal function does not exist. For example, if the send
last digit signal is not available in a national R2 signaling variant, you may set the value for req-lastfirstdigit
to 16.
The purpose of register-value is to assign values for signals requesting responses from the remote end. For
example, after you configure the register-value callingcategory command, the terminating point sends the
send calling category signal with the specified value to the originating point for the calling category.
This command applies to R2 signaling only.
Related commands: group-b enable.
NOTE:
As some national register signal coding schemes may not support all the register signals mentioned in this
section, you are recommended to use defaults unless necessary. For example, the ITU-T recommendation
is available with the send calling category signal (the callingcategory keyword) but not the send billing
category (billingcategory) signal.
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] register-value req-callingcategory 7
renew Description
Use renew to configure the values of C bit and D bit in R2 signaling.
Use undo renew to restore the default. The default value varies with R2 signaling standards in countries.
This command applies to R2 signaling only.
R2 signaling uses bits A and B to convey real status information while leaving bits C and D constant. The
values of bits C and D are national variant dependent. For example, they are fixed to 01 in most countries but
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] renew 0011
reverse Description
Use reverse to configure line signal inversion mode.
Use undo reverse to invert ABCD bits of the current line signaling whose values are ―1‖ after reverse is
executed.
This command applies to R2 signaling only.
You may configure an interface to invert the values of any ABCD bits before sending or after receiving a line
signal by replacing 0 with 1 or vice versa.
Related commands: cas and renew.
Syntax
reverse ABCD
undo reverse
View
R2 CAS view
Default level
2: System level
Parameters
ABCD: Indicates whether corresponding ABCD bits in R2 signaling need inversion. Each argument in this
command takes either of the two values: 0 for normal or 1 for inversion. The default is 0000, that is, inversion
disabled.
Examples
# Invert the values of bits B and D in R2 line signaling.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
115
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] reverse 0101
seizure-ack enable Description
Use seizure-ack enable to configure the originating point to require the terminating point to send seizure
acknowledgement signal during R2 line signaling exchange.
Use undo seizure-ack enable to restore the default.
By default, the originating point requires the terminating point to send seizure acknowledgement signal.
This command applies to R2 signaling only.
Normally, the terminating point acknowledges received seizure signals. The R2 line signaling coding
schemes in some countries however do not require the terminating point to do this. To accommodate these
schemes, you can configure the undo seizure-ack enable command, allowing the terminating point not to
acknowledge received seizure signals.
Related commands: timer dl seizure.
Syntax
seizure-ack enable
undo seizure-ack enable
View
R2 CAS view
Default level
2: System level
Parameters
None
Examples
# Disable the terminating point from sending seizure acknowledgement signals.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] undo seizure-ack enable
select-mode Description
Use select-mode to set the E1 trunk routing mode.
Use undo select-mode to restore the default.
By default, the timeslot with the smallest number is selected.
Related commands: cas and trunk-direction.
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 5
[Sysname-cas1/0:5] select-mode max
sendring ringbusy enable Description
Use sendring ringbusy enable to enable the terminating side to send busy tones to calling subscribers.
Use undo sendring ringbusy enable to disable the terminating side from sending busy tones to calling
subscribers.
By default, the terminating point sends busy tones to calling subscribers.
This command applies to R2 signaling only.
Related commands: timer ring.
Syntax
sendring ringbusy enable
undo sendring ringbusy enable
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 5
[Sysname-cas1/0:5] sendring ringbusy enable
signal-value Description
Use signal-value to configure the ABCD bit patterns of idle receive, receive seized, idle transmit, and transmit
seized signals on the digital E&M voice subscriber line.
Use undo signal-value to restore the defaults.
By default, the ABCD bit patterns of the receive idle signal and the transmit idle signal are 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.
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Examples
# Set the ABCD bit pattern to 1011 for the transmit seized signal on digital E&M subscriber line 1/0:0.
<Sysname> system-view
[Sysname] subscirber-line1/0:0
[Sysname-subscriber-line1/0:0] signal-value transmit seize 1011
special-character Description
Use special-character to configure the special characters acceptable during register signal exchange.
Use undo special-character to remove the configured special characters.
By default, no special characters are configured.
This command applies to R2 signaling only.
You may need to configure this command to accommodate to some national R2 signaling variants where
Group I forward signals can represent special characters such as pound signs (#) and asterisks (*) in
addition to digits.
NOTE:
You cannot use special-character to assign a special character different signal values.
To make sure that the device can process calls correctly, assign special characters different signal values.
Syntax
special-character character number
undo special-character character number
View
R2 CAS view
Default level
2: System level
Parameters
character: Special character, which can be a pound sign (#) or asterisk (*), A, B, C, or D.
number: Code of register signal, in the range of 11 to 16.
Examples
# Assign the pound sign (#) the register signal code 11.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] special-character # 11
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subscriber-line Description
Use subscriber-line to enter E1/T1 voice subscriber line view.
Upon creation of a TS set on an E1/T1 interface, the system automatically creates a logical voice subscriber
line numbered in the form of E1/T1 interface number:TS set number. On the voice subscriber line, you can
conveniently configure signaling and other voice functions for the corresponding E1/T1 line. On each
E1/T1 interface you can create only one TS set.
After you create a PRI group with pri-set on an E1/T1 interface, a voice subscriber line is automatically
created. This line is numbered E1 interface-number:15 on an E1 interface and T1 interface-number:23 on a
T1 interface.
Related commands: timeslot-set and pri-set.
Syntax
subscriber-line slot-number:{ ts-set-number | 15 | 23 }
View
Voice view
Default level
2: System level
Parameters
slot-number: Number of the voice subscriber line automatically created upon creation of a TS set or ISDN PRI
group.
ts-set-number: Number of the TS set that has been created.
15: Indicates the subscriber line is created for the ISDN PRI group created on an E1 interface.
23: Indicates the subscriber line is created for the ISDN PRI group created on a T1 interface.
Examples
# Enter the view of voice subscriber line 1/0:15.
<Sysname> system-view
[Sysname] subscriber-line 1/0:15
[Sysname-subscriber-line1/0:15]
tdm-clock Description
Use tdm-clock to set the TDM clock source for an E1/T1 interface.
Use undo tdm-clock to restore the default.
By default, the TDM source clock for an E1POS interface is line TDM clock, and the TDM clock source for
other E1 interfaces is the internal clock.
When digital voice E1/T1 interfaces perform TDM timeslot interchange, it is important for them to achieve
clock synchronization to prevent frame slips and bit errors.
Depending on your configurations on E1/T1 interfaces, the system adopts different clocking approaches.
When there is a subcard VCPM on the mainboard, the clock distribution principle is:
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
E1 interface view, T1 interface view
Default level
2: System level
Parameters
internal: Sets the internal crystal oscillator time division multiplexing (TDM) clock as the TDM clock source on
the E1/T1 interface. After that, the E1/T1 interface obtains clock from the crystal oscillator on the mainboard.
If it fails to do that, the interface obtains clock from the crystal oscillator on its E1/T1 card. Because SIC cards
are not available with crystal oscillator clocks, E1/T1 interfaces on SIC cards can only obtain clock from the
mainboard. The internal clock source is also referred to as master clock mode in some features.
line: Sets the line TDM clock as the TDM clock source on the E1/T1 interface. After that, the E1/T1 interface
obtains clock from the remote device through the line. The line clock source is also referred to as subordinate
(slave) clock mode in some features.
line primary: Sets the E1/T1 interface to preferably use the line TDM clock as the TDM clock source. After
that, the E1/T1 interface always attempts to use the line TDM clock prior to any other clock sources.
Examples
# Set the TDM clock source on interface E1 1/0 to line clock.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] tdm-clock line
timer dl Description
Use timer dl to configure timeouts of R2 line signals.
Use undo timer dl to restore the defaults.
This command applies to R2 signaling only.
Syntax
timer dl { answer | clear-back | clear-forward | seizure | re-answer | release-guard } time
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undo timer dl { answer | clear-back | clear-forward | seizure | re-answer | release-guard }
View
R2 CAS view
Default level
2: System level
Parameters
answer time: Timeout time in milliseconds of R2 answer signal, in the range of 100 to 120,000 with a default
of 60,000. After the originating point sends a seizure acknowledgement signal, the terminating point should
send back an answer signal within the timeout time. If the terminating point fails to send an answer signal
within the timeout time, the originating point will clear the connection. Timeout time of R2 answer signal
should be configured at both the originating point and the terminating point. The timeout time of answer
signals from the terminating point is configured at the originating point, while the timeout time of answer
signals for internal function call in a module is configured at the terminating point.
clear-back time: Timeout time in milliseconds of R2 clear-back signal, in the range of 100 to 60,000 with a
default of .10,000. After the terminating point sends a clear-back signal, it should recognize the forward
signal sent back by the originating point within the timeout time.
clear-forward time: Timeout time in milliseconds of R2 clear-forward signal configured at the originating
point, in the range of 100 to 60,000 with a default of 10,000. After the originating point sends a
clear-forward signal, the terminating point should send back a corresponding line signal, clear-back or
release guard for example, within the timeout time.
seizure time : Timeout time in milliseconds of R2 seizure signal configured at the originating point, in the
range of 100 to 5,000 with a default of 1,000. After the originating point sends a seizure signal, the
terminating point should send back a seizure acknowledgement signal within the timeout time.
re-answer time: Timeout time in milliseconds of R2 re-answer signal configured at the originating point, in
the range of 100 to 60,000 milliseconds with a default of 1,000. The originating point releases the line if it
does not receive another answer signal from the terminating point after it recognizes the clear-back signal.
release-guard time: Timeout time in milliseconds of R2 release guard signal configured at the originating
point, in the range of 100 to 60,000 with a default of 1,000. The originating point should send a release
guard signal within the timeout time after it receives a clear-back signal from the terminating point in
response to a clear-forward signal.
Examples
# Set the timeout time of R2 seizure signal to 300 milliseconds.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] timer dl seize 300
timer dtmf Description
Use timer dtmf to configure the delay from when the originating point receives a seizure acknowledgement
signal to when it starts sending DTMF signals.
Use undo timer dtmf to restore the default.
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By default, the delay is 50 milliseconds.
This command applies to R2 signaling only.
Normally, the originating point starts sending DTMF signals immediately after receiving a line seizure
acknowledgement signal. Sometimes, however, you may need to introduce a delay to accommodate to the
digit collection process on the remote PBX.
Related commands: dtmf enable.
NOTE:
Before you can configure this command, you must configure the dtmf enable command.
Syntax
timer dtmf time
undo timer dtmf
View
R2 CAS view
Default level
2: System level
Parameters
time: Delay before sending a DTMF signal in milliseconds, in the range of 50 to 10,000.
Examples
# Configure the R2 signaling to start sending DTMF signals 800 milliseconds later after receiving a seizure
acknowledgement signal.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] dtmf enable
[Sysname-cas1/0:0] timer dtmf 800
timer hold Description
Use timer hold to set the interval for sending keepalive packets.
Use undo timer hold to restore the default.
By default, the keepalive interval is 10 seconds.
Syntax
timer hold seconds
undo timer hold
View
BSV BRI interface view
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> system-view
[Sysname] interface bri 2/0
[Sysname-Bri2/0] timer hold 100
timer register-pulse persistence Description
Use timer register-pulse persistence to configure the duration of R2 register pulse signals such as A-3, A-4,
and A-6.
Use undo timer register-pulse persistence to restore the default, that is, 150 milliseconds.
By default, the duration is 150 milliseconds.
This command applies to R2 signaling only.
When the terminating point sends a backward register pulse signal, A-3 for example, the signal must persist
for a specified time period. When the originating point receives the signal, it sends back a Group II forward
signal. When the originating point recognizes the pulse signal, A4, A6, or A15, it stops sending any forward
signal, and terminates the register signal exchange.
Related commands: timer register-complete group-b.
Syntax
timer register-pulse persistence time
undo timer register-pulse persistence
View
R2 CAS view
Default level
2: System level
Parameters
persistence time: Duration in milliseconds of R2 register pulse signals, in the range of 50 to 3,000.
Examples
# Set the duration of R2 register pulse signals to 300 milliseconds.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] timer register-pulse persistence 300
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timer register-complete group-b Description
Use timer register-complete group-b to configure the timeout value of R2 group B signals. After the
terminating point switch to Group B, it should send Group B signals within this time period.
Use undo timer register-complete group-b to restore the default timeout value of R2 group B signals.
By default, the maximum time is 30,000 milliseconds.
This command applies to R2 signaling only.
Related commands: timer dl.
Syntax
timer register-complete group-b time
undo timer register-complete group-b
View
R2 CAS view
Default level
2: System level
Parameters
group-b time: Maximum time in milliseconds that the originating point waits for R2 Group B signals, in the
range of 100 to 90,000.
Examples
# Configure the maximum Group B signal exchange time to 10,000 milliseconds.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] timer register-complete group-b 10000
timer ring Description
Use timer ring to configure the duration of playing a signal tone when R2 signaling is adopted.
Use undo timer ring to restore the default duration of playing a signal tone.
By default, the duration of playing the ringback tone is 60,000 milliseconds and that of playing the busy tone
is 30,000 milliseconds.
This command applies to R2 signaling only.
Related commands: sendring.
Syntax
timer ring { ringback | ringbusy } time
undo timer ring { ringback | ringbusy }
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 0 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 0
[Sysname-cas1/0:0] timer ring ringback 10000
timeslot-set Description
Use timeslot-set to create a TS set and specify a signaling mode for it on the E1/T1 interface.
Use undo timeslot-set to remove the TS set.
By default, no TS set is configured.
You can use subscriber-line to enter subscriber line view to configure voice-related attributes only after you
create a TS set.
Related commands: subscriber-line and cas.
Syntax
timeslot-set ts-set-number timeslot-list timeslots-list signal { e&m-delay | e&m-immediate | e&m-wink |
fxo-ground | fxo-loop | fxs-ground | fxs-loop | r2 }
undo timeslot-set ts-set-number
View
E1 interface view, T1 interface view
Default level
2: System level
Parameters
ts-set-number: TS set number. For an E1 interface, the TS set number ranges from 0 to 30, and for a T1
interface, the TS set number ranges from 0 to 23.
timeslots-list: Timeslot range. Timeslots are numbered 1 through 31 for an E1 interface and 1 through 24 for
a T1 interface. TS 16 for an E1 interface (or TS24 for a T1 interface) is used to transmit control signaling.
signal: Specifies a signaling mode for the TS set, which should be consistent with that adopted by the central
office. It includes certain types of signaling:
e&m-delay: Adopts the delay start mode of digital E&M signaling.
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.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2
trunk-direction Description
Use trunk-direction to configure the R2 signal trunking direction.
Use undo trunk-direction to restore the default.
By default, bidirectional trunking applies.
This command applies to R2 signaling only.
An incoming trunk carries incoming calls but not outgoing calls while the outgoing trunk does the contrary.
A bidirectional trunk carries both incoming calls and outgoing calls.
For R2 signaling to operate normally for call connection, you need to make sure that the trunking mode is
incoming at one end of the trunk and outgoing at the other end. If both ends are using bidirectional trunking
mode, use select-mode to tune trunking policy. This is to prevent timeslot contention.
In addition, avoid using bidirectional trunking mode at one end and outgoing mode at the other end,
because this can lead to failures of outgoing calls at the end in bidirectional trunking mode.
Related commands: cas and select-mode.
Syntax
trunk-direction timeslots timeslots-list { dual | in | out }
undo trunk-direction timeslots timeslots-list
View
R2 CAS view
Default level
2: System level
Parameters
timeslots-list: Timeslot range. Timeslots are numbered 1 through 31 on an E1 interface and 1 through 24 on
a T1 interface. You may specify a single timeslot by specifying a number, a range of timeslots by specifying
a range in the form of number1-number2, or several discrete timeslots by specifying number1,
number2-number3. Examples are 1-14, 15, 17-31.
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> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2
[Sysname-e1 1/0] cas 5
[Sysname-cas1/0:5] trunk-direction timeslots 1-31 dual
ts Description
Use ts to maintain the trunk circuit of specified timeslots.
Related commands: cas.
NOTE:
The ts query command is available in R2 CAS view, digital E&M CAS view, and digital LGS CAS view.
Syntax
ts { block | open | query | reset } timeslots timeslots-list
View
R2 signaling view
Default level
2: System level
Parameters
block: Blocks the trunk circuit of specified timeslots to make it unavailable.
open: Opens the trunk circuit of specified timeslots, allowing it to carry services.
query: Queries status of the trunk circuit of specified timeslots to see whether the circuit is busy, open, or
blocked in real time.
reset: Resets the trunk circuit of specified timeslots when it cannot automatically reset. You may need to do
this if the state of an administratively blocked or opened circuit cannot recover for example.
timeslots timeslots-list: Specifies a timeslot range. Timeslots are numbered 1 through 31 for an E1 interface
and 1 through 24 for a T1 interface. You may specify a single timeslot by specifying a number, a range of
timeslots by specifying a range in the form of number1-number2, or several discrete timeslots by specifying
number1, number2-number3. Examples are 1-14, 15, 17-31.
Examples
# Reset the circuit of timeslots 1 through 15 in TS5 and query the status of the circuit of TS1 through TS31.
<Sysname> system-view
[Sysname] controller e1 1/0
[Sysname-e1 1/0] timeslot-set 5 timeslot-list 1-31 signal r2
128
[Sysname-e1 1/0] cas 5
[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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 1 voip
[Sysname-voice-dial-entity1] caller-group permit 1
caller-permit Description
Use caller-permit to configure a calling number permitted to originate calls.
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 2 pots
[Sysname-voice-dial-entity2] caller-permit 660268$
# Configure voice entity 2 to allow numbers beginning with ―20‖ to call out.
[Sysname-voice-dial-entity2] caller-permit 20
description Description
Use description to configure a subscriber group description string.
Use undo description to remove the subscriber group description string.
By default, no subscriber group description string is configured.
The description configured for a subscriber group by using description will not affect the use of the subscriber
group.
Related commands: match-template and subscriber-group.
Syntax
description text
undo description
View
Subscriber group view
Default level
2: System level
132
Parameters
text: Subscriber group description string, consisting of 1 to 80 case-insensitive characters.
Examples
# Identify subscriber group 10 as international.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] subscriber-group 10
[Sysname-voice-dial-group10] description international
dial-prefix Description
Use dial-prefix to configure a dial prefix for a voice entity.
Use undo dial-prefix to remove the configured prefix.
By default, no dial prefix is configured.
The configuration of the PBX connected to the originating router determines whether a two-stage dialing tone
is played or not.
When a voice router receives a voice call, it will compare the numbers in the match-templates of its own POTS
entities with the received called number and select one POTS entity to process the call. If a prefix is
configured, the voice router will send the prefix and dialed number together through the FXO interface.
When the number with a prefix exceeds 31 digits, only the first 31 digits are sent.
Related commands: match-template and send-number.
Syntax
dial-prefix string
undo dial-prefix
View
POTS entity view
Default level
2: System level
Parameters
string: Prefix code, a character string consisting of up to 31 characters that can include 0 through 9, comma,
#, and *. Table 28 describes these characters:
Table 28 Description of characters in the string argument
Character 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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 3 pots
[Sysname-voice-dial-entity3] dial-prefix 0
display voice subscriber-group Description
Use display subscriber-group to display the information about a subscriber group or all subscriber groups.
Syntax
display voice subscriber-group { subscriber-group-list-tag | all } [ | { begin | exclude | include }
regular-expression ]
View
Any view
Default level
2: System level
Parameters
subscriber-group-list-tag: Specifies a subscriber group ID, which ranges from 1 to 2147483647.
all: Specifies all subscriber groups.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the information about the configured subscriber groups.
<Sysname> display voice subscriber-group all
Current configuration of subscriber group 1
#
Description : <NULL>
Referenced by entities:
Type: POTS Tag: 2100
Include match templates:
Match-template: 1100..
#
134
END
Current configuration of subscriber group 2
#
Description : <NULL>
Referenced by entities:
Type: POTS Tag: 2100
Type: POTS Tag: 3100
Include match templates:
Match-template: 1200..
#
END
Table 29 Output description
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.
|: 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.
135
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the configuration information of all number substitution rule lists.
<Sysname> display voice number-substitute
Current configuration of number-substitute
#
************ NUMBER-SUBSTITUTE ************
List-tag : 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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] number-substitute 20
[Sysname-voice-dial-substitute20] dot-match right-left
first-rule Description
Use first-rule to configure the preferred number substitution rule in the current number substitution rule list.
Use undo first-rule to remove the configured preferred number substitution rule.
By default, no preferred number substitution rule is configured.
In a voice call, the system first uses the rule defined by first-rule for number substitution. If this rule fails to
apply or is not configured, it will try to apply all other rules in order until one or none of them applies.
Syntax
first-rule rule-number
undo first-rule
View
Voice number-substitute view
Default level
2: System level
Parameters
rule-number: Serial number of a number substitution rule (the serial number of a number substitution rule
configured by using the rule command), in the range of 0 to 31.
Examples
# Specify rule 4 in number substitution list 20 as the preferred rule.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] number-substitute 20
[Sysname-voice-dial-substitute20] rule 4 663 3
[Sysname-voice-dial-substitute20] first-rule 4
match-template Description
Use match-template to configure a calling number match template for a subscriber group.
Use undo match-template to delete a calling number match template or all calling number match templates
from a subscriber group.
By default, no calling number match template is configured for a subscriber group.
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:
+: Plus sign. The sign itself does not have special meanings. It only indicates that the following string is
an effective number and the number is E.164-compliant.
$: Dollar sign. When it appears at the end of the match template, it indicates the end of a calling
number, that is, only the calling number completely matching all characters before ―$‖ can originate
calls. When it does not appear, calling numbers matching the string argument can originate calls.
When it appears separately, it indicates a null calling number.
string: A string consisting of any characters of digits 0 through 9, and symbols #, *, ., !, +, %, [, ], (,
), and -.The characters in a string are described in the following table:
Table 30 Meanings of characters in a string
Character 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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] subscriber-group 2
[Sysname-voice-dial-group2] match-template 660268
max-call (voice dial program view) Description
Use max-call to configure maximum-call-connection sets.
Use undo max-call to remove the specified maximum-call-connection set or all maximum-call sets.
By default, no maximum-call-connection sets are configured.
Together with max-call in voice entity view, this command is used to limit the maximum number of call
connections of a voice entity or a set of voice entities.
Related commands: max-call (in voice entity view).
Syntax
max-call set-number max-number
undo max-call {set-number | all }
View
Voice dial program view
Parameters
set-number: Number identifying a maximum-call-connection set, in the range of 1 to 2,147,483,647. At most
256 maximum-call-connection sets can be configured.
max-number: Maximum number of call connections in a maximum-call-connection set, in the range of 0 to
120.
all: Specifies all the maximum-call-connection sets.
139
Examples
# Set the maximum number of call connections in maximum-call-connection set 1 to 5.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] max-call 1 5
max-call (voice entity view) Description
Use max-call to bind a voice entity to the maximum-call-connection set specified by the set-number argument.
Use undo max-call to remove the binding. Although you can bind each voice entity to only one
maximum-call-connection set, you can change the binding.
By default, no maximum-call-connection set is bound, that is, there is no limitation on the number of call
connections.
Related commands: max-call (in voice dial-program view).
Syntax
max-call set-number
undo max-call
View
POTS voice entity view, VoIP voice entity view, VoFR voice entity view, IVR voice entity view
Default level
2: System level
Parameters
set-number: Number identifying a maximum-call-connection set (number of the maximum-call-connection set
configured in voice dial program view), in the range of 1 to 2147483647.
Examples
# Bind voice entity 10 to maximum-call-connection set 1.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] max-call 1 5
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] max-call 1
number-match Description
Use number-match to configure a global number match mode.
Use undo number-match to restore the default number match mode.
By default, the shortest-number match mode is adopted.
Related commands: match-template and terminator.
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
Voice dial program view
Default level
2: System level
Parameters
longest: Matches the longest number.
shortest: Matches the shortest number.
Examples
# Configure the longest-number match mode.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] number-match longest
number-priority Description
Use number-priority peer enable to match a number against a voice entity match template first.
Use undo number-priority peer to restore the default.
By default, a number starting with ―*‖ or ―#‖ will first match against a service feature code.
Syntax
number-priority peer enable
undo number-priority peer
View
Voice dial program view
Default level
2: System level
Parameters
None
141
Examples
# Configure a number to first match against a voice entity match template.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] number-priority peer enable
number-substitute Description
Use number-substitute to create a number substitution rule list and enter voice number-substitute view.
Use undo number-substitute to remove a specified number substitution rule or all number substitution rule
lists.
By default, no number substitution rule list is configured.
Related commands: rule and substitute.
Syntax
number-substitute list-number
undo number-substitute { list-number | all }
View
Voice dial program view
Default level
2: System level
Parameters
list-number: Serial number of a number substitution rule list, in the range of 1 to 2147483647.
all: Specifies all number substitution rule lists.
Examples
# Enter the voice dial program view and create a number substitution rule list.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] number-substitute 1
[Sysname-voice-dial-substitute1]
priority Description
Use priority to configure the priority of a voice entity.
Use undo priority to restore the priority of a voice entity to the default.
By default, the priority level is 0.
If you have configured priority levels for voice entities and the selection priority rules (see the select-rule
commands), the router will first select the voice entity with the highest priority to initiate a call.
142
Related commands: select-rule.
Syntax
priority priority-order
undo priority
View
POTS voice entity view, VoIP voice entity view, VoFR voice entity view, IVR voice entity view
Default level
2: System level
Parameters
priority-order: Priority of a voice entity, in the range of 0 to 10. The smaller the value, the higher the priority.
Examples
# Set the priority level of voice entity 10 to 5.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] priority 5
private-line Description
Use private-line to configure the PLAR function.
Use undo private-line to disable the private line auto ring-down function.
This function is disabled by default.
This command is applicable to FXO, FXS, analog E&M interface and digital E1/T1 voice interface.
Syntax
private-line string
undo private-line
View
FXS voice subscriber line view, FXO voice subscriber line view, E&M voice subscriber line view, BSV voice
subscriber line view
Default level
2: System level
Parameters
string: E.164 telephone number of the terminating end, a string of 31 digits/characters, which can include 0
through 9, ―*‖ and ―#‖.
Examples
# Configure the private line auto ring-down function on voice subscriber line 1/0 so that 5559262 is
automatically dialed out when the subscriber picks up the phone.
<Sysname> system-view
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
Voice number-substitute view
Default level
2: System level
Parameters
all: Deletes all number substitution rules.
rule-tag: Number identifying a substitution rule, in the range of 0 to 31.
input-number: Input string of a number involved in number substitution, in the format of [ ^ ] [ + ] string [ $ ],
up to 31 characters. The signs can be explained thus:
^: Caret. The match begins with the first character of a number string. That is, the router begins with the
first character of the match string to match a user number.
+: Plus sign. The sign itself does not have special meanings. It only indicates that the following string is
an effective number and the number is E.164-compliant.
$: Dollar sign. It indicates that the last character of the match string must be matched. That is, the last
digit of a user number must match with the last character of the match string.
string: String consisting of characters such as 0 to 9, #, *, ., !, and %. Table 31 explains these
characters:
Table 31 Meanings of characters in the string argument
Character 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
argument are discarded.
For the left-right dot match rule, digits which the dots in the output-number argument correspond to are
extracted from left to right according to the number of dots in the output-number argument to replace the
dots in the output-number argument one by one. The digits that are not extracted in the input-number
argument are discarded.
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.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] number-substitute 1
# Configure number substitution rule 1 for number substitution rule list 1 thus:
Input number: 91
Output number: 1
[Sysname-voice-dial-substitute1] rule 1 ^91 1
# Configure number substitution rule 2 for number substitution rule list 1 thus:
Input number: 92
Output number: 2
[Sysname-voice-dial-substitute1] rule 2 ^92 2
# Configure number substitution rule 3 for number substitution rule list 1 thus:
Input number: 93
Output number: 3
[Sysname-voice-dial-substitute1] rule 3 ^93 3
# Configure number substitution rule 3 for number substitution rule list 1 thus:
Input number: 93
Output number: 3
Input number type: any
Output number type: International
Input numbering plan: any
Output numbering plan: telex.
[Sysname-voice-dial-substitute1] rule 3 ^93 3 number-type any international numbering-plan
any telex
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
Voice dial program view
Default level
2: System level
Parameters
1st-rule: First rule in the match order for voice entity selection. The value ranges from 1 to 4.
2nd-rule: Second rule in the match order for voice entity selection. The value ranges from 1 to 4 but differs
from that of 1st-rule.
3rd-rule: Third rule in the match order for voice entity selection. The value ranges from 1 to 4 but differs from
those of 1st-rule and 2nd-rule.
Table 36 describes the meanings of integers 1 through 4.
Table 36 Meanings of integers 1 through 4
Integer 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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] select-rule rule-order 1 2 4
select-rule search-stop Description
Use select-rule search-stop to configure the maximum number of voice entities found before a search process
stops.
Use undo select-rule search-stop to restore the default.
By default, the maximum number of voice entities found before a search process stops is 128.
The select-rule search-stop command is used to define the maximum number of qualified voice entities to be
found before a search process stops. Even if the number of voice entities meeting call requirements is greater
than max-number, the system will make call attempts to only the maximum number (max-number) of voice
entities that are matched in accordance with rules.
Related commands: select-rule rule-order and select-rule type-first.
Syntax
select-rule search-stop max-number
undo select-rule search-stop
View
Voice dial program view
Default level
2: System level
Parameters
max-number: Maximum number of voice entities found before a search process stops, in the range 1 to 128.
Examples
# Configure the maximum number of voice entities found to 5.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] select-rule search-stop 5
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
Voice dial program view
Default level
2: System level
Parameters
1st-type: Serial number of the type of the first priority, in the range of 1 to 4. Table 37 describes these values:
2nd-type: Serial number of the type of the second priority, in the range of 1 to 4. The value must be different
from that of 1st-type.
3rd-type: Serial number of the type of the third priority, in the range of 1 to 4. The value must be different from
that of 1st-type and 2nd-type.
4th-type: Serial number of the type of the fourth priority, in the range of 1 to 4. The value must be different
from that of 1st-type, 2nd-type, and 3rd-type.
Table 37 describes the meanings of these values.
Table 37 Meanings of values
Value 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.
<Sysname> system-view
[Sysname] voice-setup
150
[Sysname-voice] dial-program
[Sysname-voice-dial] select-rule type-first 2 1 3 4
select-stop Description
Use select-stop to disable the voice entity search function.
Use undo select-stop to enable the voice entity search function.
By default, the voice entity search function is enabled.
Related commands: select-rule rule-order, select-rule type-first, and select-rule search-stop.
Syntax
select-stop
undo select-stop
View
POTS voice entity view, VoIP voice entity view, VoFR voice entity view, IVR voice entity view
Default level
2: System level
Parameters
None
Examples
# Disable the voice entity search function for voice entity 10.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] select-stop
send-number Description
Use send-number to configure the number sending mode.
Use undo send-number to restore the default number sending mode.
By default, the truncate mode is used.
This command applies to only POTS voice entities. This command is used to control how to send called
numbers to PSTN. You can specify to send some digits (defined by the digit-number argument from right to
left) or all digits of called numbers. You can also specify to send truncated called numbers, the ending digits
of called numbers that match the dot (.). The dot represents the digits that match the variable part in a regular
expression. For more information, see Voice Configuration Guide.
Related commands: dot-match and match-template.
Syntax
send-number { digit-number | all | truncate }
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] send-number 6
subscriber-group Description
Use subscriber-group to create a subscriber group and enter subscriber group view, or directly enter the
subscriber group view if the subscriber group already exists.
Use undo subscriber-group to delete a subscriber group or all subscriber groups.
By default, no subscriber group is created.
At most ten subscriber groups can be configured for the system.
Related commands: description and match-template.
Syntax
subscriber-group list-number
undo subscriber-group { list-number | all }
View
Voice dial program view
Default level
2: System level
Parameters
list-number: Subscriber group ID, in the range of 1 to 2147483647.
all: Specifies all subscriber groups.
Examples
# Enter voice dial program view and create a subscriber group.
152
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] subscriber-group 1
[Sysname-voice-dial-group1]
substitute (voice subscriber line view, voice entity view) Description
Use substitute to bind a calling/called number substitution rule list to the voice subscriber line or voice entity.
Use undo substitute to remove the binding between a calling/called number substitution rule list and the
voice subscriber line or voice entity.
By default, no number substitution rule list is bound to a voice subscriber line or voice entity. That is to say,
no number substitution is performed.
Before carrying out the this command, you must first use the number-substitute list-number command to
configure a number substitution rule list in voice dial program view, and then use rule to configure rules for
the list.
According to network requirements, you can complete number substitution in the following two ways:
Before a voice entity is matched, you can use substitute in subscriber line view to substitute the
calling/called number specific to a subscriber line.
After a voice entity is matched but before a call is initiated, you can use substitute in voice entity view
to substitute a specified calling/called number.
Related commands: number-substitute and rule.
Syntax
substitute { called | calling } list-number
undo substitute { called | calling }
View
POTS voice view, VoIP voice view, VoFR voice view, subscriber line voice entity view
Default level
2: System level
Parameters
called: Applies the number substitution rule to a called number.
calling: Applies the number substitution rule to a calling number.
list-number: Serial number of a number substitution rule list configured by using the number-substitute
command), in the range of 1 to 2147483647.
Examples
# Apply number substitution rule list 6 to the called number of the voice subscriber line 1/0.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] subscriber-line1/0
[Sysname-voice-line1/0] substitute called 6
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
Voice dial program view
Default level
2: System level
Parameters
incoming-call: Binds the calling/called number of incoming calls to the number substitution rule list.
outgoing-call: Binds the calling/called number of outgoing calls to the number substitution rule list.
called: Applies the number substitution rule to a called number.
calling: Applies the number substitution rule to a calling number.
all: Specifies all number substitution rule lists.
list-number: Serial number of a number substitution rule list configured by using the number-substitute
command), in the range of 1 to 2147483647.
Examples
# Apply number substitution rule list 5 to called numbers of incoming calls.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] substitute incoming-call called 5
# Apply number substitution rule lists 5, 6, and 8 to called numbers of outgoing calls.
<Sysname> system-view
154
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] substitute outgoing-call called 5
[Sysname-voice-dial] substitute outgoing-call called 6
[Sysname-voice-dial] substitute outgoing-call called 8
terminator Description
Use terminator to configure a special character as the dial terminator for length-variable telephone numbers.
Use undo terminator to remove the dial terminator configuration.
By default, no dial terminator is configured.
If you set the argument character to # or *, and if the first character of the configured entity number is the
same as the argument character (# or *), the device will take this first character as a common number rather
than a dial terminator.
Related commands: match-template and timer.
Syntax
terminator character
undo terminator
View
Voice dial program view
Default level
2: System level
Parameters
character: Dial terminator, which can be any of 0 through 9, pound sign (#), or asterisk (*).
Examples
# Specify the pound sign (#) as the dial terminator.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] terminator #
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.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] address sip ip 3.3.3.3
# Configure the domain name of the called entity as cc.news.com for voice entity 10.
<Sysname> system-view
[Sysname] voice-setup
156
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] address sip dns cc.news.com
call-fallback Description
Use call-fallback register to enable re-registration in the case of a call failure.
Use undo call-fallback register to disable call failure-triggered re-registration.
By default, call failure-triggered re-registration is disabled.
Syntax
call-fallback register
undo call-fallback register
View
SIP client view
Default level
2: System level
Parameters
None
Examples
# Enable all failure-triggered re-registration.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] call-fallback register
crypto Description
Use crypto to reference an SSL server/client policy when TLS is used as the transport layer protocol for SIP
sessions.
Use undo crypto to remove the configuration.
By default, no SSL policy is referenced.
The SSL policies to be referenced must have been configured.
You need to first configure the TLS server and client policies, and then specify TLS as the transport layer
protocol for incoming SIP calls through the listen transport command; otherwise, no TLS requests can be
received.
If the TLS server policy or its name is modified, you need to specify TLS as the transport layer protocol again
through the transport command, and then the new policy will take effect.
If the TLS client policy or its name is modified, the new configuration will take effect for new TLS connections
and the current TLS connections still use the original policy.
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] crypto ssl-server-policy Server1
[Sysname-voice-sip] crypto ssl-client-policy Server2
display voice sip call-statistics Description
Use display voice sip call-statistics to display the statistics about all SIP calls.
Syntax
display voice sip call-statistics [ | { begin | exclude | include } regular-expression ]
View
Any view
Default level
2: System level
Parameters
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
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
Table 38 Output description
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)
Each type of message can be displayed by:
In (received)
Out (sent)
Response Message
Statistics of all SIP response messages, including 1XX, 2XX, 3XX, 4XX
(Cancel), 5XX, and 6XX
Each type of message can be displayed by:
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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the information about SIP connections over TCP.
<Sysname> display voice sip connection tcp
Conn-Id 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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display all ENUM translation rule groups.
<Sysname> display voice enum-group all
Current configuration of ENUM groups
#
enum-group 1
rule 1 preference 1 408...(8333) 555\1 cc.news.com
#
enum-group 2
rule 2 preference 3 408...(8333) 888\1 cc.news2.com
#
End
Table 39 Output description
Field 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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display SIP DNS records.
<Sysname> display voice sip dns-record
No. Host IP
1 sip1.hp.com 100.1.1.163:5060
2 sip2.8056.com 100.1.1.16:5060
Table 40 Output description
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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Query the PSTN release cause code to SIP status code mappings.
For the convenience of query, the user-defined SIP status codes are highlighted with an asterisk.
<Sysname> display voice sip reason-mapping pstn-sip
Release reason mapping of PSTN to SIP:
Index 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
Table 41 Output description
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
Table 42 Output description
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] dns-type srv
display voice sip register-state Description
Use display voice sip register-state to display status information of all user numbers to be registered on the
SIP UA.
Syntax
display voice sip register-state [ | { begin | exclude | include } regular-expression ]
View
Any view
Default level
2: System level
Parameters
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display all registration status information on the SIP UA.
<Sysname> display voice sip register-state
Number 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
Table 43 Output description
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> system-view
[Sysname] voice-setup
[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
Voice dial program view
Default level
2: System level
Parameters
group-number: Number of the ENUM translation group, ranging from 1 to 15.
all: Deletes all ENUM translation rule groups.
Examples
# Create ENUM translation rule group 1 and enter the ENUM translation view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] enum-group 1
[Sysname-voice-dial-enumgroup-1]
keepalive Description
Use keepalive to set the keepalive mode.
Use undo keepalive to restore the default keepalive mode.
Related commands: redundancy mode.
Syntax
keepalive { options [ interval seconds ] | register }
undo keepalive
View
SIP client view
Default level
2: System level
Parameters
options: Sets the keepalive mode to options.
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] keepalive options interval 30
line-check enable Description
Use line-check enable to enable checking the status of voice subscriber lines associated with POTS voice
entities.
Use undo line-check to disable checking the status of voice subscriber lines associated with POTS voice
entities.
By default, before registering numbers for a POTS voice entity, the device checks the status of the voice
subscriber line associated to the POTS voice entity. The device can send REGISTER requests for numbers only
when the status of the line is up.
Related commands: line and shutdown (voice subscriber line view).
Syntax
line-check enable
undo line-check
View
SIP client view
Default level
2: System level
Parameters
None
Examples
# Disable checking the status of voice subscriber lines associated with POTS voice entities. In other words, as
long as a POTS subscriber line is configured, the device can send REGISTER requests for numbers even if the
voice subscriber line is shut down.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] undo line-check
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> system-view
[Sysname] voice-setup
[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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] media-protocol srtp
outband sip Description
Use outband sip to configure the out-of-band SIP DTMF transmission mode.
Use undo outband sip to restore the default DTMF transmission mode.
By default, the inband DTMF transmission mode is adopted.
Syntax
outband sip
undo outband
View
POTS entity view, VoIP entity view
172
Default level
2: System level
Parameters
None
Examples
# Configure the out-of-band SIP DTMF transmission for VoIP entity 10.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 voip
[Sysname-voice-dial-entity10] address sip ip 10.1.1.2
[Sysname-voice-dial-entity10] outband sip
outbound-proxy Description
Use outbound-proxy to configure the outbound proxy server information for the SIP UA.
Use undo outbound-proxy to remove the outbound proxy server information for the SIP UA.
By default, no outbound proxy server information is configured for a SIP UA.
For more information about DTMF H.225 out-of-band transmission, DTMF H.245 out-of-band transmission,
and DTMF named NTE transmission, see Voice Configuration Guide.
Syntax
outbound-proxy { dns domain-name | ipv4 ip-address } [ port port-number ]
undo outbound-proxy { dns | ipv4 }
View
SIP client view
Default level
2: System level
Parameters
dns domain-name: Domain name of the outbound proxy server, which consists of character strings separated
by a dot, for example, aabbcc.com. Each separated string contains no more than 63 characters. A domain
name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots (.), with a maximum
length of 255 characters.
ipv4 ip-address: IPv4 address of the outbound proxy server.
port port-number: Port number of the outbound proxy server, in the range 1 to 65535.
Examples
# Configure IP address 169.54.5.10 and port number 1120 of the outbound proxy server for the SIP UA.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] outbound-proxy ipv4 169.54.5.10 port 1120
173
# Configure domain name abc.com and port number 1100 of the outbound proxy server for the SIP UA.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] outbound-proxy dns abc.com port 1100
privacy Description
Use privacy to add the P-Preferred-Identity or P-Asserted-Identity header field.
Use undo privacy to remove the configuration.
By default, neither the P-Preferred-Identity header field nor the P-Asserted-Identity header field is added.
Syntax
privacy { asserted | preferred }
undo privacy
View
SIP client view
Default level
2: System level
Parameters
asserted: Adds the P-Asserted-Identity header field. When the P-Asserted-Identity header field is added, the
Privacy header field will be added. The Privacy header field contains the caller identity presentation and
screening information, while the P-Asserted-Identity header field contains the caller identity.
preferred: Adds the P-Preferred-Identity header field. When the P-Preferred-Identity header field is added, the
Privacy header field will be added. The Privacy header field contains the caller identity presentation and
screening information, while the P-Preferred-Identity header field contains the caller identity.
Examples
# Add the P-Asserted-Identity header field.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] privacy asserted
proxy Description
Use proxy to configure proxy server information for a SIP UA.
Use undo proxy to remove the proxy server information for a SIP UA.
By default, no proxy server information is configured for SIP UA.
Syntax
proxy { dns domain-name | ipv4 ip-address } [ port port-number ]
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] proxy ipv4 169.54.5.10 port 1120
# Specify the domain name abc.com and port number 1100 for the proxy server.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] proxy dns abc.com port 1100
reason-mapping pstn Description
Use reason-mapping pstn to configure PSTN release cause code to SIP status code mappings.
Use undo reason-mapping pstn to restore the default.
By default, the PSTN release cause code to SIP status code mappings are listed in Table 44.
Table 44 Default PSTN release cause code to SIP status code mappings
PSTN release cause code
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
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PSTN release cause code
PSTN release cause description
SIP status code SIP status description
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
58 Bearer capability not
presently available! 503 Service unavailable
63 Service or option not
available, unspecified! 500 Server internal error
65 Bearer capability not
implemented! 488 Not Acceptable Here
70
Only restricted digital
information bearer
capability is available!
488 Not Acceptable Here
79 Service or option not
implemented, unspecified! 501 Not implemented
87 User not member of Closed
User Group (CUG)! 403 Forbidden
88 Incompatible destination! 503 Service unavailable
102 Recovery on timer expiry! 504 Gateway timeout
176
PSTN release cause code
PSTN release cause description
SIP status code SIP status description
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice -sip] reason-mapping pstn 17 sip 408
reason-mapping sip Description
Use reason-mapping sip to configure SIP status code to PSTN release cause code mappings.
Use undo reason-mapping sip to restore the default.
By default, the SIP status code to PSTN release cause code mappings are listed in Table 45.
Table 45 Default SIP status code to PSTN release cause code mappings
SIP status code SIP status description PSTN release cause code
PSTN release cause description
400 Bad Request 41 Temporary failure!
401 Unauthorized 21 Call rejected!
402 Payment required 21 Call rejected!
403 Forbidden 21 Call rejected!
404 Not found 1 Unallocated (unassigned)
number!
177
SIP status code SIP status description PSTN release cause code
PSTN release cause description
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
implemented, unspecified!
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
implemented, unspecified!
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
SIP status code SIP status description PSTN release cause code
PSTN release cause description
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] reason-mapping sip 486 pstn 18
register-enable Description
Use register-enable on to enable the SIP registrar.
Use register-enable off or undo register-enable to disable the SIP registrar.
By default, the SIP registrar is disabled.
Syntax
register-enable { off | on }
undo register-enable
View
SIP client view
Default level
2: System level
Parameters
on: Enables the SIP registrar.
179
off: Disables the SIP registrar.
Examples
# Enable the SIP registrar.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] register-enable on
redundancy mode Description
Use redundancy mode to set the backup mode.
Use undo redundancy mode to restore the default backup mode.
The default backup mode is parking.
Related commands: keepalive.
Syntax
redundancy mode { homing | parking }
undo redundancy mode
View
SIP client view
Default level
2: System level
Parameters
homing: Sets the backup mode to homing.
parking: Sets the backup mode to parking.
Examples
# Set the backup mode to homing.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] redundancy mode homing
registrar Description
Use registrar to configure registrar information for the SIP UA.
Use undo registrar to remove the registrar information for the SIP UA.
By default, no registrar information is configured on the SIP UA. If you execute this command without
providing the transport layer protocol type, the UDP protocol will be used during registration; if you execute
this command without providing the URL scheme, the SIP URL scheme will be used.
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> system-view
[Sysname] voice-setup
[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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] registrar dns cc.news.com port 1100 expires 120
remote-party-id Description
Use remote-party-id to add the Remote-Party-ID header field.
Use remote-party-id to remove the configuration.
By default, the Remote-Party-ID header field is not added.
Syntax
remote-party-id
undo remote-party-id
View
SIP client view
Default level
2: System level
Parameters
None
Examples
# Add the Remote-Party-ID header field.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] remote-party-id
reset voice sip connection Description
Use reset voice sip connection to clear a specified SIP connection over a specific transport layer protocol.
Syntax
reset voice sip connection { tcp | tls } id conn-id
View
User view
Default level
1: Monitor level
Parameters
tcp: Clears a SIP TCP connection.
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] enum-group 1
[Sysname-voice-dial-enum1] rule 1 preference 500 010(.{4}).* \1 beijing.gov
sip Description
Use sip to enter SIP client view.
184
Syntax
sip
View
Voice view
Default level
2: System level
Parameters
None
Examples
# Enter SIP client view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip]
sip-comp Description
Use sip-comp to configure SIP compatibility.
Use undo sip-comp to restore the default.
By default,
The destination number is obtained from the request-line, which is the start line in an SIP request
message.
The From header field contains the source address and the To header field contains the destination
address.
The compatibility options are not carried in re-INVITE requests.
The Contact header fields of REGISTER messages do not contain the dt parameter.
Syntax
sip-comp { callee | dt | from | t38 | x-parameter } *
undo sip-comp { callee | dt | from | t38 | x-parameter } *
View
SIP client view
Default level
2: System level
Parameters
callee: Configures the device to use the destination number in the To header field for sending a SIP request.
dt: Configures the Contact header fields of the REGISTER messages to contain the dt parameter. This keyword
is used when the device communicates with a VCX device.
from: Configures the device to use the address (IP address or DNS domain name) in the To header field as
the address in the From header field when sending a SIP request for interoperability with other vendors. By
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] sip-comp from
# Configure the device to use the corresponding event description in the SDP field when sending a re-INVITE
request in a fax pass-through or modem-pass-through operation.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] sip-comp x-parameter
sip-comp agent Description
Use sip-comp agent to configure the User-Agent header field in SIP request messages.
Use undo sip-comp agent to remove the configuration.
By default, the User-Agent header field in SIP request messages is not configured.
Syntax
sip-comp agent product-name product-version
undo sip-comp agent
View
SIP client view
Default level
2: System level
Parameters
agent product-name product-version: Indicates the content of the User-Agent header field in SIP request
messages. The product-name and product-version arguments respectively represent the product name and
product version of the UAC, each of which is a case-sensitive string of 1 to 31 characters, without { and }.
Examples
Set the User-Agent header field in SIP request messages to company 1.0.
<Sysname> system-view
186
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] sip-comp agent company 1.0
sip-comp server Description
Use sip-comp server to configure the Server header field in SIP response messages.
Use undo sip-comp server to remove the configuration.
By default, the Server header field in SIP response messages is not configured.
Syntax
sip-comp server product-name product-version
undo sip-comp server
View
SIP client view
Default level
2: System level
Parameters
server product-name product-version: Indicates the content of the Server header field in SIP response
messages. The product-name and product-version arguments respectively represent the product name and
product version of the UAS, each of which is a case-sensitive string of 1 to 31 characters, without { and }.
Examples
Set the Server header field in SIP response messages to company 1.1.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] sip-comp server company 1.1
sip-domain Description
Use sip-domain to configure a domain name for the SIP device.
Use undo sip-domain to remove the domain name of the SIP device.
By default, IP address, instead of domain name is used.
Syntax
sip-domain domain-name
undo sip-domain
View
SIP client view
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] sip-domain hello.com
source-bind Description
Use source-bind to bind the source IP address of SIP packets to an IPv4 address or an interface.
Use undo source-bind to remove the binding.
By default, the source IP address of SIP packets is not bound, that is, the voice gateway automatically gets an
IP address to send out SIP packets.
Syntax
source-bind { media | signal } { interface-type interface-number | ipv4 ip-address }
undo source-bind { media | signal }
View
SIP client view
Default level
2: System level
Parameters
media: Media flow.
signal: Signaling stream.
interface-type interface-number: Specifies an interface. Only Layer 3 Ethernet interfaces, GE interfaces, and
dialer interfaces are supported.
ipv4 ip-address: IPv4 address to be bound.
Examples
# Bind the IP address 1.1.1.1 to the source IP address of SIP signaling streams.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] source-bind signal ipv4 1.1.1.1
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] timer connection age tcp 6
[Sysname-voice-sip] timer connection age tls 60
timer registration retry Description
Use timer registration retry to set the interval for the voice entity or SIP trunk account to re-register with the
registrar after a registration failure.
Use undo timer registration retry to restore the default.
By default, the interval for the voice entity or SIP trunk account to re-register with the registrar after a
registration failure is 240 seconds.
Syntax
timer registration retry seconds
undo timer registration retry
View
SIP client view
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] timer registration retry 300
timer registration expires Description
Use timer registration expires to set the registration expiration time.
Use undo timer registration expires to restore the default.
By default, the registration expiration time is 3600 seconds.
Related commands: registrar, registrar server-group, timer registration divider, and timer registration
threshold.
Syntax
timer registration expires seconds
undo timer registration expires
View
SIP client view
Default level
2: System level
Parameters
Seconds: Registration expiration time in seconds, in the range 60 to 3600.
Examples
# Set the registration expiration time to 600 seconds.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] timer registration expires 600
timer registration divider Description
Use timer registration divider to set the registration percentage.
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] timer registration divider 50
timer registration threshold Description
Use timer registration threshold to set the lead time before registration.
Use undo timer registration-threshold to restore the default.
By default, the lead time before registration is 0 seconds.
Related commands: timer registration divider and timer registration expires.
Syntax
timer registration threshold seconds
undo timer registration threshold
View
SIP client view
Default level
2: System level
Parameters
seconds: Lead time (in seconds) before registration, in the range 0 to 3600.
Examples
# Set the lead time before registration to 100 seconds.
<Sysname> system-view
[Sysname] voice-setup
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] timer session-expires 1800 minimum 1000
transport Description
Use transport to specify the transport layer protocol for outgoing SIP calls.
Use undo transport to restore the default.
By default, the global transport layer protocol is UDP, and no transport layer protocol is specified for a VoIP
voice entity. If the transport layer protocol is not specified for a VoIP voice entity, the global setting is applied.
The execution of transport in SIP client view specifies the global transport layer protocol. If you want to
configure a different transport layer protocol for an individual call, you can specify the transport layer
protocol to be used in corresponding VoIP voice entity view. When the transport layer protocol configured in
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] transport tls
uri Description
Use uri to configure user information. The user information in the format user-info@domain-name is used to
send request messages.
Use undo uri to remove the user information.
By default, number@SIP-device-domain-name or number @SIP-interface-IP-address is used to send request
messages.
Related commands: sip-domain.
Syntax
uri user-info [ domain domain-name ]
undo uri
View
POTS voice entity view
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] uri user-info hello domain voice.com
url Description
Use url to specify URL scheme for SIP calls.
Use undo user to restore the default.
The execution of url in SIP client view specifies the global SIP URL scheme. If you want to configure a different
SIP URL scheme for an individual call, you can specify the SIP URL scheme in corresponding VoIP voice entity
view. When the SIP URL scheme configured in VoIP voice entity view and that configured in SIP client view
are different, the former is adopted. That is, the VoIP voice entity configuration takes precedence over global
configuration.
By default, SIP URL scheme is adopted.
You can use the SIPS scheme only when the transport layer protocol is TLS; otherwise, no session requests will
be initiated.
Syntax
url { sip | sips }
undo url
View
SIP client view, VoIP voice entity view
Default level
2: System level
Parameters
sip: Specifies SIP as the URL scheme for SIP calls.
194
sips: Specifies SIPS as the URL scheme for SIP calls.
Examples
# Specify SIPS as the global URL scheme for SIP calls.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] url sips
# Configure SIPS as the URL scheme for the SIP calls on VoIP voice entity 1000.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 1000 voip
[Sysname-voice-dial-entity1000] url sips
user Description
CAUTION:
If realm is configured on the SIP UA, make sure that the value is the same as that configured on the registrar.
Otherwise, the SIP UA will fail the authentication due to mismatch. If realm is not configured on a SIP UA, the SIP
UA will perform no realm match and consider that the value of realm configured on the registrar is trusted.
If it is necessary to configure authentication information in POTS entity view or IVR entity view, the same
authentication information is recommended for the POTS entities or IVR entities configured with the same telephone
number.
In the case of authentication, it is forbidden to execute user after the registration function is enabled because this
operation may result in registration update failures.
Use user to configure SIP authentication information.
Use undo user to restore the default.
By default, the username and password in SIP client view are VOICE-GATEWAY and VOICE-SIP, respectively,
while no SIP authentication information is configured in POTS entity view or IVR entity view.
Syntax
user username password { cipher | simple } password [ cnonce cnonce | realm realm ] *
undo user
View
SIP client view, POTS entity view, interactive voice response (IVR) entity view
Default level
2: System level
Parameters
username: Username used for registration authentication, a string of 1 to 63 case-sensitive characters. The
characters ―‖‖ and ―\‖ are invalid.
cipher: Displays the password of the current user in cipher text mode.
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] user abcd password cipher 1234
# Configure SIP authentication information in IVR entity view:
Username: abcd
Password: 1234
Display mode: cipher
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 100 ivr
[Sysname-voice-dial-entity100] user abcd password cipher 1234
wildcard-register enable Description
Use wildcard-register enable to enable fuzzy telephone number registration.
Use undo wildcard-register to disable fuzzy telephone number registration.
By default, fuzzy telephone number registration is disabled.
When configuring a match template in a POTS entity, you may use a number containing the wildcards of dot
(.) and T instead of using a standard E.164 number. After enabling fuzzy telephone number registration, the
router retains dots and substitutes asterisks (*) for Ts when sending REGISTER messages.
You can use this command only when the SIP registration function is disabled.
NOTE:
You may use fuzzy telephone number registration only when it is supported on both SIP server and
location server.
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> system-view
[Sysname] voice-setup
[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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] area-prefix 8277
[Sysname-voice-server] area-prefix 0108277
authentication Description
Use authentication to configure authentication information.
Use undo authentication to restore the default.
By default, no authentication information is configured.
If authentication is enabled on the local SIP server, users can successfully register with the local SIP server
only after authentication information is configured for them by using the authentication command.
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] register-user 1234
[Sysname-voice-server-user1234] authentication username 1234 password simple 1234
call-route Description
Use call-route to enter call route view.
Syntax
call-route
View
SIP server view
Default level
2: System level
Parameters
None
Examples
# Enter call route view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] call-route
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] call-rule-set
[Sysname-voice-server-set]
srs Description
Use srs to apply call rule set.
Use undo srs to remove the application.
By default, no call rule set is applied.
Syntax
srs tag
undo srs
View
SIP server view, register user view
Default level
2: System level
Parameters
tag: Call rule set tag, in the range of 0 to 31. The call rule set corresponding to a tag must have been
configured.
Examples
# Apply a call rule set in register user view.
200
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] call-rule-set
[Sysname-voice-server-set] service 1
[Sysname-voice-server-set-svc0] rule 1 permit outgoing 5...
[Sysname-voice-server-set-svc0] rule 2 permit outgoing 1...
[Sysname-voice-server-set-svc0] quit
[Sysname-voice-server-set] quit
[Sysname-voice-server] register-user 1000
[Sysname-voice-server-user1000] srs 1
# Apply a call rule set in sip server view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] srs 1
display voice sip-server register-user Description
Use display voice sip-server register-user to display information of registered users, including directory
number, registration status, IP address, and port number.
Syntax
display voice sip-server register-user { tag | all } [ | { begin | exclude | include } regular-expression ]
View
Any view
Default level
2: System level
Parameters
tag: Displays the information of the user with the specified tag.
all: Displays information of all users.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display information of all users.
<Sysname> display voice sip-server register-user all
user number status address
---------------------------------------------------------------------
201
1 404 online 192.168.0.98:5060
2 325 offline
3 380 online 192.168.0.57:5060
Table 46 Output description
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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the server resource information.
<Sysname> display voice sip-server resource-statistic
SIP Server state: Active
CbType 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
Table 47 Output description
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] register-user 1234
[Sysname-voice-server-user1234] expires 3700
mode Description
Use mode to configure the operation mode of the server.
Use undo mode to restore the default.
By default, the server operates in the alone mode.
You can change the operation mode of the server only when the server is disabled.
Related commands: server enable.
Syntax
mode { alive-server | alone-server }
undo mode
View
SIP server view
Default level
2: System level
Parameters
alive-server: Specifies the server to operate in the alive mode.
alone-server: Specifies the server to operate in the alone mode.
Examples
# Specify the server to operate in the alive mode.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] register-user 1234
[Sysname-voice-server-user1234] number 300
probe remote-server Description
Use probe remote-server ipv4 to configure the keepalive probe.
Use undo probe remote-server ipv4 to remove the keepalive probe.
By default, the keepalive probe is not configured.
If the local SIP server operates in the alive mode, you can configure probe remote-server ipv4 only when the
local SIP server is disabled.
Syntax
probe remote-server ipv4 ipv4-address [ port port-number ] [ keepalive time-interval ]
undo probe remote-server ipv4
View
SIP server view
Default level
2: System level
Parameters
ipv4 ipv4-address: IPv4 address of the remote server.
port port-number: Port number of the remote server, in the range of 1 to 65535. The default port number is
5060.
keepalive time-interval: Interval of sending OPTION messages to the remote server, in seconds, in the range
of 64 to 128. The default interval is 64 seconds.
205
Examples
# Configure the keepalive probe.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] register-user 1234
[Sysname-voice-server-user1234]
rule Description
Use rule to configure a call rule.
Use undo rule to remove a call rule.
By default, no call rule is configured.
Syntax
rule tag { deny | permit } { incoming | outgoing } { pattern | any }
undo rule { tag | all }
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] call-rule-set
[Sysname-voice-server-set] service 1
[Sysname-voice-server-set–svc1] 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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] call-rule-set
[Sysname-voice-server-set] service 1
[Sysname-voice-server-set-svc1]
server-bind ipv4 Description
Use server-bind ipv4 to bind the local SIP server address to the IP address of an interface on the local router.
Use undo server-bind ipv4 to remove the binding of the local SIP server address.
By default, no IP address is bound, that is, there is no local SIP server.
You can configure server- bind ipv4 only when the local SIP server is disabled.
Syntax
server-bind ipv4 ipv4-address [ port port-number ] [ expires time-interval ]
undo server-bind ipv4
View
SIP server view
Default level
2: System level
Parameters
ipv4 ipv4-address: IPv4 address. It can be the IP address of an interface on the local router, or the local
loopback address 127.0.0.1. Since the local SIP server cannot accept registrations from remote users when
the server IP address is set to 127.0.0.1, you are recommended to set the server IP address to the one of an
interface on the local router.
port port-number: Port number, in the range of 1 to 65535. The default port number is 5060.
expires time-interval: Maximum registration interval in seconds, in the range of 300 to 65535. The default
interval is 3600 seconds.
Examples
# Bind the interface address 192.168.0.92 to the address of the local SIP server.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] server-bind ipv4 192.168.0.92
server enable Description
Use server enable to enable the local SIP server.
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] server-bind ipv4 100.1.1.1
[Sysname-voice-server] server enable
sip-server Description
Use sip-server to enter sip server view.
Syntax
sip-server
View
Voice view
Default level
2: System level
Parameters
None
Examples
# Enter sip server view.
209
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server]
trunk Description
Use trunk to configure a static route entry.
Use undo trunk to delete a static route entry or all static route entries.
By default, no call route entry is configured.
Syntax
trunk tag called-number called-pattern ipv4 dest-ip-addr [ port port-number ] [ area-prefix prefix ]
undo trunk { tag | all }
View
Call route view
Default level
2: System level
Parameters
tag: Route entry tag, in the range of 0 to 31. Each tag represents a route entry. At most 32 route entries can
be configured.
called-pattern: Called number pattern, consisting of digits and dots (.). Each dot represents a character and
cannot appear before a number. This argument does not support other characters.
ipv4 dest-ip-addr: Destination IPv4 address.
area-prefix prefix: Area prefix to be added to the route entry which an internal user uses to call an external
user, consisting of 1 to 15 digits.
all: Deletes all route entries.
Examples
# Configure a static route entry, the destination address is 192.168.0.80, the called number is 1000, and the
area prefix is 5000.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[Sysname-voice-server] call-route
[Sysname-voice-server-route] trunk 20 called-number 1000 ipv4 192.168.0.80 area-prefix 5000
trusted-point Description
Use trusted-point to specify a trusted node.
Use undo trusted-point to delete a trusted node or all trusted nodes.
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-server
[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.
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sip: Specifies the SIP scheme as the URL scheme. By default, the SIP URL scheme is adopted.
sips: Specifies the SIPS scheme as the URL scheme.
Examples
# Add member server 1 to SIP server group 1, and configure the server information: set the IPv4 address of
the SIP server to 192.168.1.1, port number to 20000, and the specify TCP as the transport layer protocol for
the connection between the SIP trunk device and the SIP server.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] server-group 1
[Sysname-voice-group1] address 1 ipv4 192.168.1.1 port 20000 transport tcp
address sip server-group Description
Use address sip server-group to bind a SIP server group to a VoIP voice entity.
Use undo address sip server-group to cancel the binding between a SIP server group and a VoIP voice entity.
By default, a VoIP voice entity has no any SIP server group bound to it.
A VoIP voice entity can have only one existing SIP server group bound to it.
Related commands: address sip.
Syntax
address sip server-group group-number
undo address sip server-group
View
VoIP voice entity view
Default level
2: System level
Parameters
group-number: Specifies the index of a SIP server group, in the range 1 to 10.
Examples
# Bind SIP server group 1 to VoIP voice entity 1.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 1 voip
[Sysname-voice-dial-entity1] address sip server-group 1
assign Description
Use assign to assign the host user name or host name allocated by the ITSP to the SIP trunk account.
Use undo assign to delete the assigned host user name or host name.
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By default, no host user name or host name is assigned to a SIP trunk account.
You cannot modify or delete the host user name of a SIP trunk account when the account is enabled.
You cannot enable the registration function for a SIP trunk account before assigning a host user name for the
account.
Related commands: register enable.
Syntax
assign { contact-user user-name | host-name host-name }
undo assign { contact-user | host-name }
View
Account view
Default level
2: System level
Parameters
contact-user user-name: Host user name, a case-sensitive string of 1 to 31 characters excluding double
quotation marks (―), backslash (\), or space.
host-name host-name: Host name, a string of 1 to 255 characters, which are not case-sensitive. A host name
can include letters, digits, hyphens (-), and underscores (_), and cannot include any space.
Examples
# Assign news.com.cn as the host name to SIP trunk account 2.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-trunk account 2
[Sysname-voice-account-2] assign host-name news.com.cn
# Assign 123 as the host user name to SIP trunk account 2.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-trunk account 2
[Sysname-voice-account-2] assign contact-user 123
account enable Description
Use account enable to enable a SIP trunk account.
Use undo account enable to disable a SIP trunk account.
By default, a SIP trunk account is enabled.
Disabling a SIP trunk account that is already involved in a connection does not delete the connection. In
other words, execution of this command takes effect on the next calling of this account.
Syntax
account enable
undo account enable
214
View
Account view
Default level
2: System level
Parameters
None
Examples
# Disable SIP trunk account 2.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-trunk account 2
[Sysname-voice-account-2] undo account enable
bind sip-trunk account Description
Use bind sip-trunk account to bind a SIP trunk account to a VoIP voice entity.
Use undo bind sip-trunk account to cancel the binding between a SIP trunk account and a VoIP voice entity.
By default, a VoIP voice entity has no any SIP trunk account bound to it.
Only an existing SIP trunk account can be bound to a VoIP voice entity.
Canceling the binding between a VoIP voice entity and a SIP trunk account that is already involved in a
connection does not delete the connection. In other words, execution of this command takes effect on the next
calling of this account.
Syntax
bind sip-trunk account account-index
undo bind sip-trunk account
View
VoIP voice entity view
Default level
2: System level
Parameters
account-index: Index of the SIP trunk account to be bound to a VoIP voice entity, in the range 1 to 16.
Examples
# Bind SIP trunk account 1 to VoIP voice entity 1.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 1 voip
[Sysname-voice-dial-entity1] bind sip-trunk account 1
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codec transparent Description
Use codec transparent to enable codec transparent transmission.
Use undo codec transparent to restore the default.
By default, codec transparent transmission is disabled, and the SIP trunk device participates in media
negotiation between two parties.
Enable codec transparent transmission on the VoIP voice entities attached to the internal and external
networks.
Syntax
codec transparent
undo codec transparent
View
VoIP voice entity view
Default level
2: System level
Parameters
None
Examples
# Enable codec transparent transmission.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 1 voip
[Sysname-voice-dial-entity1] codec transparent
description Description
Use description to configure the description for a SIP server group.
Use undo description to delete the description for a SIP server group.
By default, a SIP server group has no description configured.
Syntax
description text
undo description
View
Server group view
Default level
2: System level
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> system-view
[Sysname] voice-setup
[Sysname-voice] server-group 1
[Sysname-voice-group-1] description ITSPA
display voice sip-trunk account Description
Use display voice sip-trunk account to display SIP trunk account information.
Syntax
display voice sip-trunk account [ | { begin | exclude | include } regular-expression ]
View
Any view
Default level
1: Monitor level
Parameters
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display SIP trunk account information.
<Sysname> display voice sip-trunk account
ID 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
Table 48 Output description
Field Description
ID SIP trunk account index
User Host user name
Group SIP server group index
Server Address or domain name of the registrar
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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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the details of SIP server group 1.
<Sysname> display voice server-group 1
The information of server group 1
Group name: ITSPA
Description: ITSP 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
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Hot swap mode: Disabled
Table 49 Output description
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.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] 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> system-view
[Sysname] voice-setup
[Sysname-voice] server-group 1
[Sysname-voice-group-1] hot-swap enable
keepalive Description
Use keepalive to enable the keep-alive function and set the keep-alive mode for a SIP server group.
Use undo keepalive to disable the keep-alive function for a SIP server group.
By default, the keep-alive function for a SIP server group is disabled.
With the keep-alive function enabled, the SIP trunk device selects the current server according to the detect
result and the redundancy mode. If the keep-alive function is disabled, the current server is always the one
with the highest priority in the SIP server group.
Related commands: redundancy mode.
Syntax
keepalive { options [ interval seconds ] | register }
undo keepalive
View
Server group view
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> system-view
[Sysname] voice-setup
[Sysname-voice] server-group 1
[Sysname-voice-group-1] keepalive register
match source host-prefix Description
Use match source host-prefix to match a source host name prefix for a VoIP voice entity.
Use undo match source host-prefix to delete the call match rule that specifies the prefix of source host name.
By default, no prefix of the source host name is specified as a call match rule for a VoIP voice entity. In other
words, all source host names can be matched.
The specified prefix of source host name is used to match against the source host names of calls. If the
INVITE message received by the SIP trunk device carries the Remote-Party-ID header, the calling
information is abstracted from this header field; if the INVITE message received by the SIP trunk device
carries the Privacy header, the source host name is abstracted from the P-Asserted-Identity or
P-Preferred-Identity header field; if the INVITE message received by the SIP trunk device does not carry
any of the above mentioned three header fields, the host name in the From header field of the INVITE
message is used as the source host name.
You can specify only one source host name prefix based match rule for a VoIP voice entity. If you
execute match source host-prefix multiple times, the new configuration overwrites the existing one.
Syntax
match source host-prefix host-prefix
undo match source host-prefix
View
VoIP voice entity view
Default level
2: System level
Parameters
Host-prefix: Source host name prefix. The value consists of 1 to 31 characters, which are not case-sensitive
and can include letters, digits, underlines (_), hyphens (-), asterisk (*), and dots (.). An asterisk represents a
character string of any length, for example, t*m can match the source host names tom, tim, and so on.
221
Examples
# Specify that calls with a source host name starting with Bil are permitted on VoIP voice entity 1.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 1 voip
[Sysname-voice-dial-entity1] match source host-prefix bil
match destination host-prefix Description
Use match destination host-prefix to match a destination host name prefix for a VoIP voice entity.
Use undo match destination host-prefix to delete the call match rule that specifies the prefix of destination
host name.
By default, no prefix of destination host name is specified as a call match rule for a VoIP voice entity. In other
words, all destination host names can be matched.
The specified prefix of destination host name is used to match against the destination host names of
calls. The host name in the To header field of an INVITE message received by the SIP trunk device is
used as the destination host name.
You can specify only one destination host name prefix based match rule for a VoIP voice entity. If you
execute match destination host-prefix multiple times, the new configuration overwrites the existing one.
Syntax
match destination host-prefix host-prefix
undo match destination host-prefix
View
VoIP voice entity view
Default level
2: System level
Parameters
Host-prefix: Destination host name prefix. The value consists of 1 to 31 characters, which are not
case-sensitive and can include letters, digits, underlines (_), hyphens (-), asterisk (*), and dots (.). An asterisk
represents a character string of any length, for example, b*y can match the destination host names boy,
boundary, and so on.
Examples
# Specify that calls with a destination host name starting with ali are permitted on VoIP voice entity 1.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 3 voip
[Sysname-voice-dial-entity3] match destination host-prefix ali
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
VoIP voice entity view
Default level
2: System level
Parameters
ipv4 ip-address: Source IP address. The value must be dotted and can include dots (.), multiplication signs (x),
asterisks (*), and digits, where x represents any number between 0 and 9, * represents any number between
0 and 255, and x and * can appear multiple times in one source IP address. Fuzzy matching is supported.
For example, 100.1.x.3 indicates any IP address between 100.1.0.3 and 100.1.9.3, and 192.*.*.* indicates
any IP address between 192.0.0.1 and 192.255.255.255.
dns dns-name: Domain name. A domain name is not case-insensitive and can include letters, digits, hyphens
(-), underscores (_), asterisk (*), and dots (.), with a maximum length of 255 characters. If you provide this
parameter, the specified domain name is used to match against the source addresses of calls, and a
whole-word match is considered a match. For example, if the domain name is configured as sohu, sohu.com
is not a match. However, fuzzy matching is supported. An asterisk represents a character string of any length,
for example, i*n can match the source addresses ilison, iverson, inn, and so on.
server-group group-number: SIP server group index, in the range 1 to 10.
Examples
# Specify that calls with a source IP address in the range 100.1.1.1 to 100.1.1.255 are permitted on VoIP voice
entity 3.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 3 voip
[Sysname-voice-dial-entity3] match source address ipv4 100.1.1.*
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] proxy server-group 5
registrar server-group Description
Use registrar server-group to associate a SIP trunk account with a SIP server group for registration.
Use undo registrar server-group to delete the association between a SIP trunk account and a SIP server
group.
By default, a SIP trunk account has no SIP server group associated for registration.
The specified SIP server group must exist. One SIP trunk account can be associated with only one SIP server
group.
A SIP trunk account registration cannot be enabled if the account is not associated with any SIP server group.
Related commands: register enable and timer registration expires.
Syntax
registrar server-group group-number [ expires seconds ]
undo registrar server-group
View
Account view
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-trunk account 1
[Sysname-voice-account-1] registrar server-group 2 expires 300
register enable Description
Use register enable to enable the registration function for a SIP trunk account.
Use undo register enable to disable the registration function for a SIP trunk account.
By default, the registration function for a SIP trunk account is disabled.
To enable the registration function for a SIP trunk account, you need to assign it with a host user name or
associate it with a SIP server group.
When the registration function for a SIP trunk account is enabled, you cannot change its host user name or
associated SIP server group.
Related commands: assign and registrar server-group.
Syntax
register enable
undo register enable
View
Account view
Default level
2: System level
Parameters
None
Examples
# Assign 123 as the host name for SIP trunk account 2, and associate SIP trunk account 2 with SIP server
group 2. Then, enable the registration function for the SIP trunk account.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-trunk account 2
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] redundancy mode homing
server-group Description
Use server-group to create a SIP server group and enter server group view. If the created server group
already exits, use this command to enter server group view.
Use undo server-group to delete one or all SIP server groups.
A SIP server group that is bound to a SIP trunk account or a VoIP voice entity cannot be deleted.
The undo server-group all can be executed successfully only when all SIP server groups are not bound to any
SIP trunk account or a VoIP voice entity.
Syntax
server-group group-number
undo server-group { group-number | all }
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> system-view
[Sysname] voice-setup
[Sysname-voice] server-group 1
[Sysname-voice-group-1]
sip-trunk account Description
Use sip-trunk account to create a SIP trunk account and enter SIP trunk account view. If the created SIP trunk
account already exits, use this command to enter SIP trunk account view.
Use undo sip-trunk account to delete one or all SIP trunk accounts.
A SIP trunk account that is bound to a SIP server group or a VoIP voice entity cannot be deleted.
The undo sip-trunk account all command can be executed successfully only when all SIP trunk accounts are
not bound to any SIP server group or a VoIP voice entity.
Related commands: bind sip trunk-account.
Syntax
sip-trunk account account-index
undo sip-trunk account { account-index | all }
View
Voice view
Default level
2: System level
Parameters
account account-index: SIP trunk account index, in the range 1 to 16.
all: Specifies all SIP trunk accounts.
Examples
# Create SIP trunk account 2 and enter its view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip-trunk account 2
[Sysname-voice-account-2]
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip-trunk enable
user Description
Use user to configure the authentication user name and password for a SIP trunk account.
Use undo user to delete the configured authentication user name and password for a SIP trunk account.
By default, a SIP trunk account has no authentication user name or password.
Syntax
user username password { cipher | simple } password
undo user
View
Account view
Default level
2: System level
Parameters
username: SIP trunk account username used for registration authentication, a case-sensitive string of 1 to 63
characters. The characters ―‖‖ and ―\‖ are invalid.
simple: Displays the password of the current account in plain text.
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> system-view
[Sysname] voice-setup
[Sysname-voice] 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> system-view
[Sysname] voice-setup
[Sysname-voice] backup-rule loose
call-forwarding no-reply enable Description
Use call-forwarding no-reply enable to enable call forwarding no reply.
Use undo call-forwarding no-reply enable to restore the default.
By default, call forwarding no reply is disabled.
Related commands: call-forwarding on-busy enable, call-forwarding unavailable enable, call-forwarding
unconditional enable, and call-forwarding priority.
NOTE:
This command applies only to FXS voice subscriber lines.
Syntax
call-forwarding no-reply enable forward-number number
undo call-forwarding no-reply enable
230
View
Voice subscriber line view
Default level
2: System level
Parameters
forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31
digits, 0 through 9.
Examples
# Enable call forwarding no reply for voice subscriber line 1/0 and set the forwarded-to number to
12345678.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-forwarding no-reply enable forward-number 12345678
call-forwarding on-busy enable Description
Use call-forwarding on-busy enable to enable call forwarding busy.
Use undo call-forwarding on-busy enable to restore the default.
By default, call forwarding busy is disabled.
Related commands: call-forwarding no-reply enable, call-forwarding unavailable enable, call-forwarding
unconditional enable, and call-forwarding priority.
NOTE:
This command applies only to the FXS voice subscriber line.
Syntax
call-forwarding on-busy enable forward-number number
undo call-forwarding on-busy enable
View
Voice subscriber line view
Default level
2: System level
Parameters
forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31
digits, 0 through 9.
Examples
# Enable call forwarding busy for voice subscriber line 1/0 and set the forwarded-to number to 12345678.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-forwarding on-busy enable forward-number 12345678
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:
This command applies only to the FXS voice subscriber line.
By default, the priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3 respectively. The
smaller the value is, the higher the priority level is. When you change the priority level of a feature, make sure that
different features have different priority levels.
Syntax
call-forwarding priority level
undo call-forwarding priority
View
Voice subscriber line view
Default level
2: System level
Parameters
level: Call forwarding priority level, in the range of 1 to 3. The smaller the value, the higher the priority.
Examples
# Configure the call forwarding priority level of 1 for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-forwarding priority 1
call-forwarding unavailable enable Description
Use call-forwarding unavailable enable to enable call forwarding unavailable.
Use undo call-forwarding unavailable enable to restore the default.
By default, call forwarding unavailable is disabled.
Related commands: call-forwarding on-busy enable, call-forwarding no-reply enable, call-forwarding
unconditional enable, and call-forwarding priority.
NOTE:
This command applies only to the FXS voice subscriber line.
232
Syntax
call-forwarding unavailable enable forward-number number
undo call-forwarding unavailable enable
View
Voice subscriber line view
Default level
2: System level
Parameters
forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31
digits, 0 through 9.
Examples
# Enable call forwarding unavailable for voice subscriber line 1/0 and set the forwarded-to number to
12345678.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-forwarding unavailable enable forward-number 12345678
call-forwarding unconditional enable Description
Use call-forwarding unconditional enable to enable call forwarding unconditional.
Use undo call-forwarding unconditional enable to restore the default.
By default, call forwarding unconditional is disabled.
Related commands: call-forwarding on-busy enable, call-forwarding no-reply enable, call-forwarding
unavailable enable, and call-forwarding priority.
NOTE:
This command applies only to the FXS voice subscriber line.
Syntax
call-forwarding unconditional enable forward-number number
undo call-forwarding unconditional enable
View
Voice subscriber line view
Default level
2: System level
Parameters
forward-number number: Specifies a forwarded-to number in the E.164 format, which is a string of 1 to 31
digits, 0 through 9.
233
Examples
# Enable call forwarding unconditional for voice subscriber line 1/0 and set the forwarded-to number to
12345678.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-forwarding unconditional enable forward-number 12345678
call-hold enable Description
Use call-hold enable to enable call hold.
Use undo call-hold enable to disable call hold.
By default, call hold is disabled.
NOTE:
This command is only applicable to the FXS voice subscriber line.
Syntax
call-hold enable
undo call-hold enable
View
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable the call hold feature for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-hold enable
call-hold-format Description
Use call-hold-format to configure a tone playing mode for call hold.
Use undo call-hold-format to restore the default.
By default, the tone playing mode is inactive, that is, the silent mode.
Syntax
call-hold-format { inactive | sendonly [ media-play media-id ] }
undo call-hold-format
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> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] media-file g729r8
[Sysname-voice-ivr-g729r8] set-media 1919 cf:/g729/music.wav
[Sysname-voice-ivr-g729r8] quit
[Sysname-voice-ivr] quit
[Sysname-voice] call-hold-format sendonly media-play 1919
call-transfer enable Description
Use call-transfer enable to enable call transfer.
Use undo call- transfer enable to disable call transfer.
By default, call transfer is disabled.
Related commands: call-transfer start-delay.
NOTE:
This command applies only to the FXS voice subscriber line.
Call hold must be enabled before call transfer.
Syntax
call-transfer enable
undo call- transfer enable
View
Voice subscriber line view
Default level
2: System level
235
Parameters
None
Examples
# Enable call transfer for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-transfer enable
call-transfer start-delay Description
Use call-transfer start-delay to configure a call transfer start delay.
Use undo call-transfer start-delay to restore the default call transfer start delay.
By default, the call transfer start delay is 3 seconds.
Related commands: call-transfer enable.
NOTE:
This command applies only to the FXS voice subscriber line.
Call hold must be enabled before call transfer.
Syntax
call-transfer start-delay number
undo call-transfer start-delay
View
Voice subscriber line view
Default level
2: System level
Parameters
number: Call transfer start delay in seconds, in the range of 2 to 5.
Examples
# Set the call transfer start delay to 2 seconds for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-transfer start-delay 2
call-waiting Description
Use call-waiting to configure parameters for a call waiting tone.
Use undo call-waiting to restore the default.
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
Voice subscriber line view
Default level
2: System level
Parameters
cwi-count number: Number of a call waiting tone play times, in the range of 1 to 5.
cwi-duration length: Number of tones played at one time, in the range of 1 to 3.
cwi-interval length: Interval for playing a call waiting tone in seconds, in the range of 10 to 30.
Examples
# Specify a call waiting tone to be played twice for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-waiting cwi-count 2
call-waiting enable Description
Use call-waiting enable to enable call waiting.
Use undo call-waiting enable to disable call waiting.
By default, call waiting is disabled.
Related commands: call-waiting and call-waiting priority.
Syntax
call-waiting enable
undo call-waiting enable
View
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable call waiting for voice subscriber line 1/0.
237
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-waiting enable
call-waiting priority Description
Use call-waiting priority to configure a priority level for call waiting.
Use undo call-waiting priority to restore the default.
By default, the call waiting priority level is 3.
Related commands: call-waiting and call-waiting priority.
NOTE:
By default, the priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3 respectively.
The smaller the value is, the higher the priority level is. When you change the priority level of a feature,
make sure that different features have different priority levels.
Syntax
call-waiting priority level
undo call-waiting priority
View
Voice subscriber line view
Default level
2: System level
Parameters
level: Call waiting priority level, in the range of 1 to 3. The smaller the value, the higher the priority.
Examples
# Configure the call waiting priority level of 1 for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] call-waiting priority 1
conference enable Description
Use conference enable to enable the three-party conference function for a voice subscriber line.
Use undo conference enable to restore the default.
By default, the three-party conference function is disabled.
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
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable the three-party conference function for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] conference enable
dialin-restriction enable Description
Use dialin-restriction enable to enable incoming call barring for a voice subscriber line.
Use undo dialin-restriction enable to restore the default.
By default, incoming call barring is disabled.
Syntax
dialin-restriction enable
undo dialin-restriction enable
View
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable incoming call barring for voice subscriber line 1/0.
<Sysname> system-view
239
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] dialin-restriction enable
dialout-restriction enable Description
Use dialout-restriction enable to enable outgoing call barring for a voice subscriber line. Use undo
dialout-restriction enable to restore the default.
By default, outgoing call barring is disabled.
Syntax
dialout-restriction enable password { cipher | simple } password
undo dialout-restriction enable
View
Voice subscriber line view
Default level
2: System level
Parameters
password: Sets a password for outgoing call barring.
cipher password: Specifies a password in either plain or cipher text mode, and displays it in cipher test mode.
A password in plain text must be a string of 1 to 4 decimal digits. A password in cipher text mode must be
a string of 24 characters.
simple password: Specifies a password in plain text mode and displays it in plain text mode. A password in
plain text mode must be a string of 1 to 4 decimal digits.
Examples
# Enable outgoing call barring for voice subscriber line 1/0 and set a password to 1234 in plain text mode.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] dialout-restriction enable password simple 1234
display voice sip subscribe-state Description
Use display voice sip subscribe-state to display the information of subscription, including phone numbers,
subscription server address, effective time, and subscription state.
display voice sip subscribe-state [ | { begin | exclude | include } regular-expression ]
View
Any view
Default level
1: Monitor level
240
Parameters
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the information of subscription.
<Sysname> display voice sip subscribe-state
Number Server Address Expires Status
+-----------------------------------------------------------------------+
1515 100.1.1.101:5060 3600 online
Table 50 Output description
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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display all information of MWI.
<Sysname> display voice ss mwi all
Message Waiting Indication information:
---------------------------------------------------------------------
MWI type: Bind
MWI server: 100.1.1.101 port:5060
MWI expires: 3600
---------------------------------------------------------------------
Number: 1515
Messages-Waiting: Yes
Voicemail: 1/3(1/2)
Total: 4(3)
Table 51 Output description
Field Description
MWI 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:
This command applies only to the FXS voice subscriber line.
The Feature service indicates the service that is used together with the VCX. When you need to interact with the VCX
by using telephone keys, you need to adopt out-of-band NTE transmission to send the DTMF digits to the VCX. The
execution of feature permit does not enable out-of-band NTE transmission, and you need to execute outband nte
on the called entity to enable it. For more information about the out-of-band NTE transmission, see Voice Configuration Guide.
Syntax
feature { deny | permit }
undo feature
View
Voice subscriber line view
Default level
2: System level
Parameters
deny: Disables the setting of the Feature service.
permit: Enables the setting of the Feature service.
Examples
# Enable the setting of the Feature service for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] feature permit
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
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable hunt group for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] hunt-group enable
hunt-group priority Description
Use hunt-group priority to configure a priority level for hunt group.
Use undo hunt-group priority to restore the default.
By default, the hunt group priority level is 1.
Related commands: hunt-group enable.
NOTE:
By default, the priority levels for hunt group, call forwarding, and call waiting are 1, 2, and 3 respectively.
The smaller the value is, the higher the priority level is. When you change the priority level of a feature,
make sure that different features have different priority levels.
Syntax
hunt-group priority level
244
undo hunt-group priority
View
Voice subscriber line view
Default level
2: System level
Parameters
level: Hunt group priority level, in the range of 1 to 3. The smaller the value, the higher the priority.
Examples
# Configure the hunt group priority level of 2 for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] hunt-group priority 2
joined-conference enable Description
CAUTION:
Enabling the three-party conference service in active participation mode will invalidate the local call
identification function (if configured).
Use joined-conference enable to enable the three-party conference service in active participation mode for
a voice subscriber line.
Use undo joined-conference enable to restore the default.
By default, the three-party conference in active participation mode is disabled.
Related commands: distinguish-localtalk (in the chapter ―Voice entity configuration commands‖).
Syntax
joined-conference enable
undo joined-conference enable
View
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable the three-party conference service in active participation mode for voice subscriber line 1/0.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] joined-conference enable
245
mwi enable Description
Use mwi enable to enable MWI.
Use undo mwi enable to disable MWI.
By default, MWI is disabled.
NOTE:
This command is only applicable to the FXS voice subscriber line.
Syntax
mwi enable
undo mwi enable
View
Voice subscriber line view
Default level
2: System level
Parameters
None
Examples
# Enable MWI for voice subscriber line 3/0.
<Sysname> system-view
[Sysname] subscriber-line 3/0
[Sysname-subscriber-line3/0] mwi enable
mwi tone-duration Description
Use mwi tone-duration to configure the duration of playing the message waiting tone.
Use undo mwi tone-duration to restore the default.
By default, the duration of the message waiting tone is 2 seconds.
NOTE:
This command is only applicable to the FXS voice subscriber line.
Syntax
mwi tone-duration length
undo mwi tone-duration
View
Voice subscriber line view
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> system-view
[Sysname] subscriber-line 3/0
[Sysname-subscriber-line3/0] mwi tone-duration 4
mwi-server Description
Use mwi-server to configure the related information of the voice mailbox server.
Use undo mwi-server to remove the configurations.
By default, no voice mailbox server information is configured.
Before specifying the transport layer protocol with the mwi-server command, you need to configure the same
transport layer protocol with the listen transport command; otherwise, no subscription request can be
initiated.
Before specifying TLS as the transport layer protocol with the mwi-server command, , you need to reference
an SSL client policy with the crypto command; otherwise, no subscription request can be initiated.
Syntax
mwi-server { dns domain-name | ipv4 ip-address } [ expires seconds ] [ port port-number ] [ retry seconds ]
[ tcp | tls ] [ scheme { sip | sips } ] { bind | no-bind { loose | strict } }
undo mwi-server
View
SIP client view
Default level
2: System level
Parameters
dns domain-name: Specifies the domain name of the voice mailbox server, which consists of character
strings separated by a dot (for example, aabbcc.com). Each separated string contains no more than 63
characters. A domain name can include case-insensitive letters, digits, hyphens (-), underscores (_), and dots
(.), with a maximum length of 255 characters.
ipv4 ip-address: IP address of the voice mailbox server.
expires seconds: Effective time of the subscription in seconds, which is in the range 10 to 72000, and
defaults to 3600.
port port-number: Port number of the voice mailbox server, which is in the range 1 to 65535.
retry seconds: Subscription retry interval in seconds, which is in the range 10 to 7200, and defaults to 120.
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> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] mwi-server ipv4 100.1.1.101 port 5060 expires 7200 retry 180 bind
# Configure the domain name of the MWI server as cc.hp.com, port number as 5060, subscription effective
time as 3600 seconds, subscription retry interval as 240 seconds, and the binding mode as bind.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] mwi-server dns cc.hp.com port 5060 expires 3600 retry 240 bind
# Configure the IP address of the voice mailbox server as 192.168.0.88, transport layer protocol as TCP, and
the binding mode as bind.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] sip
[Sysname-voice-sip] mwi-server ipv4 192.168.0.88 tcp bind
timer called-hookon-delay Description
Use timer called-hookon-delay to enable calling party control and set the on-hook delay time of the called
party.
Use undo timer called-hookon-delay to restore the default.
By default, calling party control is disabled, that is, the on-hook delay of the called party is set to 0.
248
Syntax
timer called-hookon-delay seconds
undo timer called-hookon-delay
View
Analog FXS voice subscriber line view
Default level
2: System level
Parameters
seconds: Specifies the on-hook delay of the called party, in seconds, in the range of 0 to 90.
Examples
# Enable calling party control on voice subscriber line 1/0 and set the on-hook delay time of the called party
to 90 seconds.
<Sysname> system-view
[Sysname] subscriber-line 1/0
[Sysname-subscriber-line1/0] timer called-hookon-delay 90
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> system-view
[Sysname] controller e1 1/0
[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> system-view
[Sysname] call-watch rule 1 local interface ethernet 1/1
# Create monitor group 2 and associate it with track object ID 1.
<Sysname> system-view
[Sysname] call-watch rule 2 remote track 1
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 }
regular-expression ]
View
Any view
Default level
1: Monitor level
Parameters
controller controller-type controller-number: Specifies an E1 or T1 interface by its type and number.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display information of all call-watch groups associated with an E1/T1 interface.
<Sysname> display call-watch status
Controller E1 1/0 : UP
Call-Watch mode : hard
Call-Watch rule 1 remote track 1, state is INVALID
Controller E1 1/1 : Watch-Out
Call-Watch mode : soft
Call-Watch rule 2 local interface Ethernet1/1, network is DOWN
Table 52 Output description
Field 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
Voice dial program 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.
9600: Negotiates the baud rate first in accordance with the V.29 fax protocol. The maximum
transmission rate to 9600 bps.
14400: Negotiates the baud rate first in accordance with the V.17 fax protocol. The maximum
transmission rate to 14,400 bps.
disable: Disables fax forwarding.( If the call control protocol is SIP, this keyword disables forwarding of
private T.38 and standard T.38 faxes only.)
voice: Sets the fax rate to the allowed maximum voice speed for different codec protocols.
cng-switch enable: Enables CNG fax switchover.
ecm: Enables the fax error correction mode. It is disabled by default.
level level: Specifies the fax signal level in dBm (in the range of –60 to –3). The default value is –15.
local-train threshold threshold: Specifies the threshold percentage of fax local training (in the range of 0 to
100). The default value is 10.
nsf-on: Enables NSF message transmission. It is disabled by default.
protocol: Specifies the transport protocol of the fax. By default, the T.38 fax protocol is applied. Both
hb-redundancy number and lb-redundancy number default to 0.
standard-t38: Adopts the standard T.38 (UDP) fax protocol, which supports SIP-T.38 protocol.
pcm: Enables the pass-through mode.
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the statistics about the FoIP module.
<Sysname> display voice fax statistics
Statistics about Fax Session:
{
Total : 0
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_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
Failure : 0
FAX_VOFR_STANDARD_SWITCH: 0
FAX_VOFR_FRF11_TRUNK : 0
FAX_VOFR_FRF11_SWITCH : 0
256
FAX_VOFR_MOTOROLA : 0
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_VOFR_STANDARD_SWITCH: 0
FAX_VOFR_FRF11_TRUNK : 0
FAX_VOFR_FRF11_SWITCH : 0
FAX_VOFR_MOTOROLA : 0
FAX_VOIP_STDT38 : 0
FAX_VOIP_T38 : 0
}
Statistics about using fax baudrate:
{
V27 2400 : 0
V27 4800 : 0
V29 7200 : 0
V29 9600 : 0
V17 7200 : 0
V17 9600 : 0
V17 12000: 0
V17 14400: 0
}
Statistics about using ECM or Non-ECM mode:
{
ECM : 0
Non-ECM: 0
}
Statistics 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
}
Table 53 Output description
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
Statistics of the number of connections released in the
case that the DP does not start modulation within the
specified time
WAIT_DP_END_DEMODULATE_TIMEOUT
Statistics of the number of connections released in the
case that the DP does stop demodulation within the
specified time
WAIT_DP_END_MODULATE_TIMEOUT
Statistics of the number of connections released in the
case that the DP does not stop modulation within the
specified time
WAIT_FRAMEACK_TIMEOUT
Statistics of the number of connections released in the
case that no Frame ACK message is received from the DP
within the specified time
WAIT_T30MSG_PSTN_TIMEOUT
Statistics of the number of connections released in the
case that no T.30 message is received from PSTN within
the specified time
WAIT_T30MSG_IP_TIMEOUT
Statistics of the number of connections released in the
case that no T.30 message is received from the IP network
within the specified time
SPOOL_TIME_OVER
Statistics of the number of connections released in the
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.
9600: Negotiates the fax baud rate first in accordance with the V.29 fax protocol. The maximum fax baud
rate is 9,600 bps.
14400: Negotiates the fax baud rate first in accordance with the V.17 fax protocol. The maximum fax baud
rate is 14,400 bps.
disable: Disables the fax function. (If the call control protocol is SIP, private T.38 and standard T.38 faxes are
disabled.)
voice: Finalizes the allowed maximum fax baud rate first in accordance with voice encoding/decoding
protocols.
If G.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.
If G.726 is adopted, the fax baud rate is 14,400 bps and the fax protocol is V.17.
If G.729 is adopted, the fax baud rate is 7,200 bps and the fax protocol is V.29.
Examples
# Configure the gateway to negotiate the fax rate in accordance with the V.29 fax protocol.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 4 pots
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
POTS entity view, VoIP entity view
Default level
2: System level
Parameters
None
Examples
# Enable the CNG fax switchover function.
<sysmane> system-view
[sysname] voice-setup
[sysname-voice] dial-program
[sysname-voice-dial] entity 100 pots
[sysname-voice-dial-entity100] fax cng-switch enable
fax ecm Description
Use fax ecm to configure the ECM mode for the fax.
Use undo fax ecm to restore the default.
By default, the ECM mode is not used on the gateway.
The fax ecm command is used to perform the forced restriction on the gateway. Only when the fax terminals
on both sides support the ECM mode and the gateway uses the ECM mode, the ECM mode will be selected.
You must enable the ECM mode for the POTS and VoIP entities of the fax sender and receiver in the ECM
mode.
NOTE:
The configuration of fax ecm in voice entity view is invalid for the FRF.11 trunk mode.
Syntax
fax ecm
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 4 pots
[Sysname-voice-dial-entity4] fax ecm
fax level Description
Use fax level to configure the transmit energy level of a gateway carrier.
Use undo fax level to restore the default.
By default, the transmit energy level of a gateway carrier is –15 dBm.
Usually, the default transmit energy level of a gateway carrier is acceptable. If fax still cannot be sent when
other configurations are correct, try to adjust the transmit energy level.
Syntax
fax level level
undo fax level
View
POTS entity view, VoIP entity view, VoFR entity view
Default level
2: System level
Parameters
level: Level of the energy transmitted by a gateway carrier, the transmit energy level attenuation value in dBm,
in the range of –60 to –3. The greater the level value is, the higher the energy is. The smaller the level value
is, the greater the attenuation is.
Examples
# Configure the transmit energy level of the gateway carrier to –20 dBm.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 4 pots
[Sysname-voice-dial-entity4] fax level -20
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
POTS entity view, VoIP entity view, VoFR entity view
Default level
2: System level
Parameters
threshold: Local training threshold in percentage, in the range of 0 to 100.
Examples
# Configure the percentage of the local training threshold to 20.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] fax local-train threshold 20
fax nsf-on Description
Use fax nsf-on common to configure the signal transmission mode of fax faculty as a nonstandard mode.
Use undo fax nsf-on to restore the default transmission mode.
By default, the standard signal transmission mode of fax faculty is adopted.
Syntax
fax nsf-on
262
undo fax nsf-on
View
POTS entity view, VoIP entity view, VoFR entity view
Default level
2: System level
Parameters
None
Examples
# Configure a nonstandard faculty for fax signal transmission.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] fax nsf-on
fax protocol Description
Use fax protocol to configure the type of protocol used for fax communication with other devices.
Use undo fax protocol to restore the default type of protocol used for fax communication with other devices.
By default, T.38 negotiation mode is used for fax.
If the call control protocol is SIP, this command can be used only for the originator of the fax request. When
a fax request is originated, the fax type is decided according to the configurations. The receiver of the fax
request responds to the originator based on the type of the fax request, and then establishes a fax call.
Low-speed data refers to the V.21 command data, while high-speed data refers to the TCF and image
data.
To communicate with leading fax terminals in the industry, the standard T.38 protocol must be selected.
Likewise, to communicate with other fax terminals supporting a T.38 protocol, the T.38 protocol must be
adopted. As the leading devices do not support local training mode for fax, the point-to-point training
mode must be adopted in order to implement interworking with the leading devices in the industry.
Increasing the number of redundant packets will improve reliability of network transmission and reduce
packet loss ratio. A great amount of redundant packets, however, can increase bandwidth consumption
to a great extent and thereby, in the case of low bandwidth, affect the fax quality seriously. Therefore,
the number of redundant packets should be selected properly according to the network bandwidth.
The pass-through mode is subject to such factors as loss of packet, jitter and delay, so the clocks on both
communication sides must be kept synchronized. At present, only G.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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 4 pots
[Sysname-voice-dial-entity4] fax protocol t38 hb-redundancy 2
fax train-mode Description
Use fax train-mode to configure the fax training mode.
Use undo fax train-mode to restore the default.
By default, the point-to-point mode is adopted.
NOTE:
VoFR entities only support the PPP training mode.
Syntax
fax train-mode { local | ppp }
undo fax train-mode
View
POTS entity view, VoIP entity view, VoFR entity view
Default level
2: System level
264
Parameters
local: Adopts the local training mode.
ppp: Adopts the ppp training mode.
Examples
# Configure the local training mode for the gateway.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] fax train-mode local
modem compatible-param Description
Use modem compatible-param to configure the value of NTE payload type for the NTE-compatible switching
mode.
Use undo modem compatible-param to restore the default.
By default, the value of the NTE payload type is 100.
This command is valid only for the NTE-compatible switching mode.
Related commands: modem protocol pcm.
Syntax
modem compatible-param payload-type
undo modem compatible-param
View
POTS entity view, VoIP entity view
Default level
2: System level
Parameters
payload-type: Value of the NTE payload type for the NTE-compatible switching mode, in the range of 98 to
120.
Examples
# Set the NET payload type to 99 for the NTE-compatible switching mode.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] modem compatible-param 99
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
POTS entity view, VoIP entity view
Default level
2: System level
Parameters
standard: Uses Re-Invite switching for Modem pass-through.
nte-compatible: Uses NTE-compatible switching for Modem pass-through.
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 pots
[Sysname-voice-dial-entity10] modem protocol pcm nte-compatible g711alaw
reset voice fax statistics Description
Use reset voice fax statistics to clear IP fax statistics.
Syntax
reset voice fax statistics
View
User view
Default level
2: System level
Parameters
None
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> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 call
[Sysname-voice-ivr-node1] call-normal terminator #
# Configure node 1 to receive a normal secondary-call number by matching the length of the number.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 call
[Sysname-voice-ivr-node1] call-normal length 7
# Configure node 1 to receive a normal secondary-call number by matching the number.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
268
[Sysname-voice-ivr] node 1 call
[Sysname-voice-ivr-node1] call-normal matching
description Description
Use description to configure the description string for the node.
Use undo description to remove the configuration.
By default, no node description string is configured.
Syntax
description text
undo description
View
Call node view, Jump node view, Service node view
Default level
2: System level
Parameters
text: Node description string of 1 to 80 case-sensitive characters. Spaces are permitted.
Examples
# Configure the description string for the Jump node.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 jump
[Sysname-voice-ivr-node1] description first-node
display voice ivr call-info Description
Use display voice ivr call-info to display IVR call information.
Syntax
display voice ivr call-info [ | { begin | exclude | include } regular-expression ]
View
Any view
Default level
2: System level
Parameters
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
269
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display IVR call information.
<Sysname> display voice ivr call-info
Index 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
Table 54 Output description
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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
270
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the IVR playing information.
<Sysname> display voice ivr media-play
Index 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
Table 55 Output description
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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display IVR media resource information.
<Sysname> display voice ivr media-source
Codec Media-Id source Size (Bytes) Read-Number Cache-Number
--------------------------------------------------------------------------
g729r8 1000 cfa0:/wav/g7 69304 1 1
29r8/0.wav
Table 56 Output description
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 }
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View
Voice dial program view
Default level
2: System level
Parameters
entity-number: Number of an IVR voice entity, in the range 1 to 2147483647.
all: All types of voice entities, including VoIP, POTS, VoFR, and IVR voice entities.
ivr: Indicates that the voice entity type is IVR.
Examples
#Create IVR voice entity 100 and enter voice entity view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 100 ivr
extension Description
Use extension to configure an extension secondary-call for a node. You can configure at most ten extension
secondary-call numbers for a Call node.
Use undo extension to remove the configuration.
By default, no extension secondary-call number is configured.
Syntax
extension extension-number call corresponding-number
undo extension extension-number
View
Call node view
Default level
2: System level
Parameters
extension-number: Number to be input by the subscriber, a string of 1-31 characters, including 0 through 9,
pound sign (#), or asterisk (*).
corresponding-number: Extension number, a string of 1-31 characters, including 0 through 9, pound sign (#),
or asterisk (*).
Examples
# Configure an extension secondary-call for node 1: when the subscriber dials the number 0, node 1
executes the secondary-call to the number 5000.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
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[Sysname-voice-ivr] node 1 call
[Sysname-voice-ivr-node1] extension 0 call 5000
input-error Description
Use input-error to configure the processing method for handling subscriber input errors for a node.
Use undo input-error to remove the configuration.
By default, no input error processing method is configured for a node.
Syntax
input-error { end-call | goto-pre-node | goto-node node-id } [ media-play media-id [ play-times ] | repeat
repeat-times ] *
undo input-error
View
Call node view, Jump node view
Default level
2: System level
Parameters
end-call: Terminates the call when the maximum number of input errors is reached.
goto-pre-node: Return to the previous node when the maximum number of input errors is reached.
goto-node node-id: Jumps to a specified node when the maximum number of input errors is reached.
media-play media-id: Specifies the ID of the media resource file to be played after an input error and before
the node is executed again, in the range 0 to 2147483647.
play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1.
repeat repeat-times: Specifies the maximum number of input errors. After an input error occurs, the node will
be executed again. When the maximum number of input errors is reached, the system processes according
to the configured method. The value of the repeat-times argument ranges from 0 to 255 and defaults to 3.
Examples
# Configure the processing method for handling subscriber input errors for Jump node 1:
The node should terminate a call after the maximum number of input errors is reached.
The media resource ID is 1000.
The node plays voice prompts six times.
The maximum number of input errors is five.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 jump
[Sysname-voice-ivr-node1] input-error end-call media-play 1000 6 repeat 5
# Configure the processing method for handling subscriber input errors for Jump node 1:
274
The node should return to the previous node after the maximum number of times permitted for inputting
errors is reached.
The media resource ID is 1001.
The node plays voice prompts only once.
The maximum number of input errors is three.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 jump
[Sysname-voice-ivr-node1] input-error goto-pre-node media-play 1001 1 repeat 3
# Configure the processing method for handling subscriber input errors for Jump node 1:
The node should jump to node 20 after the maximum number of times permitted for inputting errors is
reached.
The media resource ID is 1002.
The node plays voice prompts three times.
The maximum number of input errors is five.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr ] node 1 jump
[Sysname-voice-ivr-node1 ] input-error goto-node 20 media-play 1002 3 repeat 5
ivr-input-error Description
Use ivr-input-error to configure processing method for handling subscriber input errors globally.
Use undo ivr-input-error to restore the default.
By default, the maximum number of input errors is three. The system does not play voice prompts for input
errors and terminates the call after the maximum number of input errors is reached.
Syntax
ivr-input-error { media-play media-id [ play-times ] | repeat repeat-times } *
undo ivr-input-error
View
IVR management view
Default level
2: System level
Parameters
media-play media-id: Specifies the ID of the media resource file to be played after an input error occurs and
before the node is executed again. The value ranges from 0 to 2147483647.
play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1.
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repeat repeat-times: Specifies the maximum number of input errors. After an input error occurs, the node will
be executed again. When the maximum number of input errors is reached, the system terminates the call. The
value of the repeat-times argument ranges from 0 to 255 and defaults to 3.
Examples
# Configure the global processing method for handling subscriber input errors:
The media resource ID is 1002.
The node plays voice prompts twice.
The maximum number of input errors is five.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] ivr-input-error media-play 10002 2 repeat 5
ivr-root Description
Use ivr-root to specify the root node (the first node to be executed) of an IVR voice entity.
Use undo ivr-root to remove the configuration.
By default, the root node is not configured for an IVR voice entity.
Syntax
ivr-root node-id
undo ivr-root
View
IVR voice entity view
Default level
2: System level
Parameters
node-id: Specifies the ID of the root node, in the range 1 to 256.
Examples
# Configure the root node of IVR voice entity 100.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 100 ivr
[Sysname-voice-dial-entity100] ivr-root 1
ivr-system Description
Use ivr-system to enter IVR management view.
276
Syntax
ivr-system
View
Voice view
Default level
2: System level
Parameters
Node
Examples
# Enter IVR management view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr]
ivr-timeout Description
Use ivr-timeout to configure the IVR global input-timeout processing method.
Use undo ivr-timeout to restore the default.
By default, the timeout time is 10 seconds, and the maximum timeout times are three. The system does not
play voice prompts for input timeouts and terminates the call after the maximum number of times is reached.
Syntax
ivr-timeout { expires seconds | media-play media-id [ play-times ] | repeat repeat-times } *
undo ivr-timeout
View
IVR management view
Default level
2: System level
Parameters
expires seconds: Specifies the timeout time. The value ranges from 1 to 255 and defaults to 10, in seconds.
media-play media-id: Specifies the ID of the media resource file to be played after an input timeout occurs
and before the node is executed again. The value ranges from 0 to 2147483647.
play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1.
repeat repeat-times: Specifies the maximum number of input timeouts. After an input timeout occurs, the node
will be executed again. When the maximum number of input timeouts is reached, the system terminates the
call. The value of the repeat-times argument ranges from 0 to 255 and defaults to 3.
Examples
# Configure the global input timeout processing method:
277
The timeout time is 20 seconds.
The media resource ID is 100001.
The node plays voice prompts only once.
The maximum number of timeout is twice
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] ivr-timeout expires 20 media-play 100001 1 repeat 2
media-file Description
Use media-file to enter voice media resource management view.
Related commands: ivr-system and set-media.
Syntax
media-file { 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> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] media-file g729r8
[Sysname-voice-ivr-g729r8]
media-play Description
Use media-play to specify the audio file that will be played to the subscriber when the node is waiting for the
subscriber to press keys.
Use undo media-play to restore the default.
By default, the audio file that will be played to the subscriber when the node is waiting for the subscriber to
press keys is not configured.
278
Syntax
media-play media-id [ play-times ] [ force ]
undo media-play
View
Call node view, Jump node view
Default level
2: System level
Parameters
media-id: Media resource file ID, in the range 0 to 2147483647.
play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1.
force: Specifies that the subscriber can press the key only after the play of voice prompts is finished,
otherwise, subscriber input is considered invalid. By default, the force keyword is not specified, that is,.
subscriber input is valid during voice prompt display.
Examples
# Specify the node to play the audio file 10000 three times to the subscriber when waiting for the subscriber
to press keys.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 jump
[Sysname-voice-ivr-node1] media-play 10000 3 force
node Description
Use node to create an IVR voice entity node or enter the view of an existing node.
Use undo node to delete a specified or all IVR nodes.
Syntax
node node-id { call | jump | service }
undo node { node-id | all }
View
IVR management view
Default level
2: System level
Parameters
node-id: Specifies the node ID, in the range 1 to 256.
call: Creates a Call node, which executes a secondary-call after the subscriber inputs a number.
jump: Creates a Jump node, which jumps to another node according to the input of the subscriber.
service: Creates a Service node, which executes various operations, such as playing audio files, jumping,
executing immediate secondary-call, terminating a call, and playing voice prompts.
279
all: All types of nodes.
Examples
# Create Jump node 1 and enter its view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 jump
[Sysname-voice-ivr-node1]
operation Description
Use operation to configure function for a Service node.
Use undo operation to remove the configuration.
By default, no function is configured for a Service node.
If an executed function is to jump to another node or to terminate a call, the rest one or two functions will not
be executed.
Related commands: select-rule operation-order.
Syntax
operation number { call-immediate call-number | end-call | goto-node node-id | goto-pre-node |
media-play media-id [ play-times ] }
undo operation number
View
Service node view
Default level
2: System level
Parameters
number: Specifies the serial number of the configured function, in the range 1 to 3.
call-immediate call-number: Indicates immediate secondary-call. The call-number argument represents the
phone number of the secondary-call.
end-call: Terminates a call.
goto-node node-id: Jumps to a specified node. The node-id argument represents the node ID, in the range 1
to 256.
goto-pre-node: Returns to the previous node.
media-play media-id: Specifies the media resource file ID, in the range 0 to 2147483647.
play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1.
Examples
# Configure Service node 1.
<Sysname> system-view
[Sysname] voice-setup
280
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 service
[Sysname-voice-ivr-node1] operation 1 end-call
select-rule operation-order Description
Use select-rule operation-order to specify the execution order of the configured functions.
Use undo select-rule operation-order to restore the default.
By default, the execution order is select-rule operation-order 1 2 3.
Related commands: operation.
Syntax
select-rule operation-order 1st-operation 2nd-operation 3rd-operation
undo select-rule operation-order
View
Service node view
Default level
2: System level
Parameters
1st-operation: Specifies the serial number of the function to be executed first. The value ranges from 1 to 3.
2nd-operation: Specifies the serial number of the function to be executed secondly. The value ranges from 1
to 3, and cannot be the same as the value of 1st-operation.
3rd-operation: Specifies the serial number of the function to be executed thirdly. The value ranges from 1 to
3, and cannot be the same as the value of 1st-operation and 2nd-operation.
Examples
# Specify the execution order of the configured functions for node 1 as 1->3->2.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 service
[Sysname-voice-ivr-node1] select-rule operation-order 1 3 2
set-media Description
Use set-media to specify a media resource ID for a media resource file. Each codec can be configured with
up to 256 media resource IDs.
Use undo set-media to remove the configuration.
By default, no customized media ID is specified for a media resource file.
Related commands: media-file.
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> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] media-file g729r8
[Sysname-voice-ivr-g729r8] set-media 10001 file cfa0:/g729/ring.wav
timeout Description
Use timeout to configure the input timeout processing method for an IVR node.
Use undo timeout to remove the configuration.
By default, no input timeout processing method is configured for an IVR node.
Syntax
timeout { end-call | goto-pre-node | goto-node node-id } [ expires seconds | media-play media-id
[ play-times ] | repeat repeat-times ] *
undo timeout
View
Call node view, Jump node view
Default level
2: System level
Parameters
end-call: Terminates the call when the maximum number of input timeouts is reached.
goto-pre-node: Return to the previous node when the maximum number of input timeouts is reached.
goto-node node-id: Jumps to a specified node when the maximum number of input timeouts is reached. The
value ranges from 1 to 256.
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.
play-times: Specifies the times for playing voice prompts. The value ranges from 1 to 255 and defaults to 1.
repeat repeat-times: Specifies the maximum number of input timeouts. After an input timeout occurs, the node
will be executed again. When the maximum number of input timeouts is reached, the system terminates the
call. The value of the repeat-times argument ranges from 0 to 255 and defaults to 3.
Examples
# Configure the input timeout processing method for Jump node 1:
The node should terminate the call after the maximum number of times permitted for input timeouts is
reached.
The maximum number of input timeouts is three.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 jump
[Sysname-voice-ivr-node1] timeout end-call repeat 3
user-input Description
Use user-input to configure the node to execute the jump operation based on the input of the subscriber.
Use undo user-input to remove the configuration.
By default, no jump operation is configured.
You can configure up to 12 operations for a Jump node.
Syntax
user-input character { end-call | goto-node node-id | goto-pre-node }
undo user-input character
View
Jump node view
Default level
2: System level
Parameters
character: Input of the subscriber. The value can be any of 0 through 9, pound sign (#), or asterisk (*).
end-call: Terminates the call.
goto-node node-id: Jumps to the specified node. The value node-id ranges from 1 to 256.
goto-pre-node: Return to the previous node.
Examples
# Configure the node to terminate the call if the subscriber presses 0.
<Sysname> system-view
283
[Sysname] voice-setup
[Sysname-voice] ivr-system
[Sysname-voice-ivr] node 1 jump
[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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 4 vofr
[Sysname-voice-dial-entity4] match-template 12345
[Sysname-voice-dial-entity4] address vofr-dynamic serial1/0 100
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 vofr
[Sysname-voice-dial-entity10] call-mode static
cid select-mode Description
Use cid select-mode to configure the CID selection mode which the originating side of a VoFR call adopts.
Use undo cid select-mode to restore the default.
By default, CIDs are cyclically selected in descending order.
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> system-view
[Sysname] interface serial 1/0
[Sysname-Serial1/0] fr dlci 100
[Sysname-fr-dlci-100] cid select-mode min-poll
display fr vofr-info Description
Use display fr vofr-info to display the FRF.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 }
regular-expression ]
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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
287
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the FRF.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
Table 57 Output description
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
Voice dial program 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.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 vofr
outband vofr Description
Use outband vofr to configure the out-of-band DTMF transmission mode.
Use undo outband to restore the default.
By default, the inband DTMF transmission mode is adopted.
Syntax
outband vofr
undo outband
View
VoFR entity view
Default level
2: System level
Parameters
None
Examples
# Configure the out-of-band DTMF transmission mode for VoFR entity 10.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 vofr
[Sysname-voice-dial-entity10] outband vofr
seq-number Description
Use seq-number to configure the VoFR packets sent by the local voice gateway to carry a sequence number.
Use undo seq-number to restore the default.
By default, the VoFR packets sent by the local voice gateway do not carry any sequence number.
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 vofr
[Sysname-voice-dial-entity10] seq-number
timestamp Description
Use timestamp to configure VoFR packets sent by the local voice gateway to carry a timestamp.
Use undo timestamp to restore the default.
By default, the VoFR packets sent by the local voice gateway do not carry any timestamp.
Syntax
timestamp
undo timestamp
View
VoFR entity view
Default level
2: System level
290
Parameters
None
Examples
# Configure voice packets sent by VoFR entity 10 to carry a timestamp.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 10 vofr
[Sysname-voice-dial-entity10] timestamp
trunk-id Description
Use trunk-id to configure a PSTN-dialed number in the FRF.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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] entity 2222 vofr
[Sysname-voice-dial-entity2222] call-mode static
[Sysname-voice-dial-entity2222] trunk-id 3333
voice bandwidth Description
Use voice bandwidth to reserve a VoFR voice bandwidth.
Use undo voice bandwidth to remove the reserved bandwidth.
By default, no bandwidth is reserved for voice.
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> system-view
[Sysname] fr class test1
[Sysname-fr-class-test1] voice bandwidth 8000 reserved
vofr Description
Use vofr to configure a VoFR operation mode for a DLCI.
Use undo vofr to restore the default.
By default, no VoFR operation mode is configured.
If the VoFR operation mode is set to Motorola-compatible and the call mode is set to static (FRF.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.
Related commands: call-mode.
Syntax
vofr { huawei-compatible [ dce | dte ] | motorola-compatible [ dce | dte ] | nonstandard-compatible
signal-channel ccid-no data-channel dcid-no [ keepalive ] }
undo vofr
View
Interface DLCI view
Default level
2: System level
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> system-view
[Sysname] interface serial 1/0
[Sysname-Serial1/0] link-protocol fr ietf
[Sysname-Serial1/0] fr dlci 100
[Sysname-fr-dlci-Serial1/0-100] vofr nonstandard-compatible signal-channel 4 data-channel 5
keepalive
# Set the call control protocol on DLCI 200 to Huawei-compatible (DTE).
<Sysname> system-view
[Sysname] interface serial 1/0
[Sysname-Serial1/0] link-protocol fr ietf
[Sysname-Serial1/0] fr dlci 200
[Sysname-fr-dlci-Serial1/0-100] vofr huawei-compatible dte
vofr frf11-timer Description
Use vofr 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.
Related commands: call-mode.
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> system-view
[Sysname] voice-setup
[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> system-view
[Sysname] voice-setup
[Sysname-voice] aaa-client
[Sysname-voice-aaa]
accounting Description
Use accounting to enable the RADIUS accounting function for users who dial some access number.
Use undo accounting to disable the RADIUS accounting function.
By default, the RADIUS accounting function is disabled for users who dial access numbers.
On one voice gateway, the RADIUS accounting function for one-stage dialing users (who dial a called
number to originate a call after picking up the phone) differs from that for two-stage dialing users (who first
dial an access number and then a called number to originate a call after picking up the phone). This
command is only applicable to an access number, two-stage dialing users. With the RADIUS accounting
function enabled, the RADIUS server will perform accounting for all users who use this access number. With
the function disabled, the RADIUS server will not perform accounting for users who dial the access number.
Related commands: gw-access-number, acct-method, and accounting-did.
Syntax
accounting
undo accounting
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] accounting
# Disable the RADIUS accounting function for users who dial the access number 17909.
[Sysname-voice-dial-anum17909] undo accounting
accounting-did Description
Use accounting-did to enable the RADIUS accounting function for all one-stage dialing users.
Use undo accounting-did to disable the RADIUS accounting function.
By default, the RADIUS accounting function is disabled for all one-stage dialing users.
On one voice gateway, the RADIUS accounting for one-stage dialing users is separated from that for
two-stage dialing users. This command is applicable to only one-stage dialing users. With this function
enabled, the RADIUS server will perform RADIUS accounting for all calls originated by one-stage dialing
users. With this function disabled, the RADIUS server will not perform accounting for any calls originated by
one-stage dialing users.
Related commands: acct-method and accounting.
Syntax
accounting-did
undo accounting-did
View
Voice AAA client view
Default level
2: System level
Parameters
None
Examples
# Enable the accounting function for all one-stage dialing users.
<Sysname> system-view
296
[Sysname] voice-setup
[Sysname-voice] aaa-client
[Sysname-voice-aaa] accounting-did
# Disable the accounting function for all one-stage dialing users.
[Sysname-voice-aaa] undo accounting-did
acct-method Description
Use acct-method to configure an accounting method for the RADIUS client.
Use undo acct-method to restore the default.
By default, the accounting method is start-no-ack.
Related commands: accounting and accounting-did.
Syntax
acct-method { start-ack | start-no-ack | stop-only }
undo acct-method
View
Voice AAA client view
Default level
2: System level
Parameters
start-ack: When the call setup begins, the voice gateway sends an Accounting-Start request to the RADIUS
server. However, the voice gateway must receive an Accounting_Start acknowledgment from the RADIUS
server before connecting the call. After the call ends, the voice gateway sends an Accounting_Stop request
to the RADIUS server, and releases the call upon receiving an Accounting_Stop acknowledgment.
start-no-ack: When the call setup begins, the voice gateway sends an Accounting_Start request to the
RADIUS server, and directly connects the call without waiting for an Accounting_Start acknowledgment. If
the voice gateway receives an Accounting_Start unacknowledgment from the RADIUS server after the call is
connected, it immediately releases the call. After the call ends, the voice gateway sends an Accounting_Stop
request to the RADIUS server and releases the call only after it receives an Accounting_Stop
acknowledgment.
stop-only: The voice gateway sends an Accounting_Stop request to the RADIUS server only when the call
ends, and it releases the call only after receiving an Accounting_Stop acknowledgment.
Examples
# Set the accounting method to start-ack.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] aaa-client
[Sysname-voice-aaa] acct-method start-ack
# Restore the default accounting method.
[Sysname-voice-aaa] undo acct-method
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] authentication
# Disable the authentication function for users who dial the access number 17909.
[Sysname-voice-dial-anum17909] undo authentication
authentication-did Description
Use authentication-did to enable the authentication function for all one-stage dialing users.
Use undo authentication-did to disable the authentication function.
By default, the authentication function is disabled for all one-stage dialing users.
This command is applicable to only one-stage dialing users, instead of two-stage dialing users.
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
Voice AAA client view
Default level
2: System level
Parameters
None
Examples
# Enable the authentication function for one-stage dialing users.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] aaa-client
[Sysname-voice-aaa] authentication-did
authorization Description
Use authorization to enable the authorization function for users who dial some access number.
Use undo authorization to disable the authorization function.
By default, the authorization function is disabled for users who dial access numbers.
With this function enabled, called numbers will be sent to the RADIUS server for authorization after users who
dial some access number to make IP calls pass authentication.
You must enable the authentication function (by using the authentication command) before the authorization
function. Otherwise, authorization is unavailable.
Related commands: gw-access-number and authentication.
Syntax
authorization
undo authorization
View
Access number view
299
Default level
2: System level
Parameters
None
Examples
# Enable the authorization function for users who dial the access number 17909.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] authentication
[Sysname-voice-dial-anum17909] authorization
# Disable the authorization function for users who dial the access number 17909.
[Sysname-voice-dial-anum17909] undo authorization
authorization-did Description
Use authorization-did to enable the authentication function for all one-stage dialing users.
Use undo authorization-did to disable the authorization function for all one-stage dialing users.
By default, the authorization function is disabled for all one-stage dialing users.
This command is applicable to only one-stage dialing users, instead of two-stage dialing users. With this
function enabled, called numbers will be sent to the RADIUS server for authorization after users who dial
some access number to make IP calls pass authentication.
You must enable the authentication function before the authorization function. Otherwise, authorization-did
is unavailable.
Related commands: authentication-did.
Syntax
authorization-did
undo authorization-did
View
Voice AAA client view
Default level
2: System level
Parameters
None
Examples
# Enable the authorization function for one-stage dialing users.
<Sysname> system-view
[Sysname] voice-setup
300
[Sysname-voice] aaa-client
[Sysname-voice-aaa] authentication-did
[Sysname-voice-aaa] authorization-did
# Disable the authorization function for one-stage dialing users.
[Sysname-voice-aaa] undo authorization-did
callednumber receive-method Description
Use callednumber receive-method to configure the method of collecting digits of a called number.
Use undo callednumber receive-method to restore the default.
By default, users need to press the dial terminator # after dialing all digits of a called number.
This command is applicable to both the one-stage dialing process and two-stage dialing process. In the
terminator mode, the voice gateway can immediately originate a call only after users dial a called number
and press the dial terminator #, and otherwise, the voice gateway will not originate a call until timeout. In the
immediate mode, the voice gateway can originate a call immediately it collects all digits of a called number,
without waiting for users to press the dial terminator #. The immediate mode simplifies users‘ operations.
Related commands: gw-access-number.
Syntax
callednumber receive-method { immediate | terminator }
undo callednumber receive-method
View
Access number view
Default level
2: System level
Parameters
immediate: Specifies the voice gateway to originate a call immediately it collects all digits of a called
number.
terminator: Specifies users to press the dial terminator # after dialing a called number.
Examples
# Set the method of collecting digits of called numbers to immediate for the access number 17909.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] callednumber receive-method immediate
# Restore the default method of collecting digits of called numbers for the access number 17909.
[Sysname-voice-dial-anum17909] undo callednumber receive-method
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] process-config cardnumber
[Sysname-voice-dial-anum17909] card-digit 10
cdr Description
Use cdr to configure a rule for saving CDRs.
Use undo cdr to restore the default saving rule, and undo cdr all to restore the values of buffer, duration, and
threshold all to the defaults.
The voice gateway will save a certain amount of CDRs according to the configured rule. When you set the
number of CDRs that can be saved or the lifetime of CDRs, the voice gateway will judge whether the existing
CDRs will be deleted. If so, the voice gateway will prompt for confirmation and determine whether to validate
the configuration according to your confirmation.
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
Voice AAA client view
Default level
2: System level
Parameters
buffer size-number: Specifies the number of CDRs that can be saved in the buffer. The size-number argument
ranges from 0 to 500, with a default of 50. The value ―0‖ indicates that no CDR can be saved.
duration time-length: Specifies the lifetime of CDRs in seconds. The time-length argument ranges from 0 to
2,147,483,647, with a default of 86,400. The value ―0‖ indicates that no CDR can be saved.
threshold percentage: Specifies the alarm threshold in percentage for CDRs. When the percentage of the
saved CDRs in the total CDRs that can be saved in the buffer reaches the alarm threshold, the voice gateway
will generate alarm information once. The percentage argument ranges from 0 to 100, with a default of 80.
The value ―0‖ indicates that no alarm information will be output.
Examples
# Set the number of CDRs that can be saved to 400.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] aaa-client
[Sysname-voice-aaa] cdr buffer 400
# Set the lifetime of CDRs to 10 hours.
[Sysname-voice-aaa] cdr duration 36000
# Set the alarm threshold for CDRs to 10%.
[Sysname-voice-aaa] cdr threshold 10
display voice access-number Description
Use display voice access-number to display the configuration information and access numbers in voice AAA
client view.
The information displayed includes:
Accounting method
Enabling or disabling of the authentication, authorization, and accounting functions for one-stage
dialing users
Rule for saving CDRs
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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display the configuration information and access numbers.
<Sysname> display voice access-number
AAA configuration :
accounting-method = 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
Table 58 Output description
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.
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display call records by calling number.
<Sysname> display voice call-history-record callingnumber 4000
Call records of voice RADIUS:
#
CallRecord [ 0 ]:
CallReference = 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
Table 59 Output description
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
|: Filters command output by specifying a regular expression. For more information about regular
expressions, see Fundamentals Configuration Guide.
begin: Displays the first line that matches the specified regular expression and all lines that follow.
exclude: Displays all lines that do not match the specified regular expression.
include: Displays all lines that match the specified regular expression.
regular-expression: Specifies a regular expression, a case-sensitive string of 1 to 256 characters.
Examples
# Display statistics of messages exchanged between the voice RADIUS module, CMC module, and AAA
module.
<Sysname> display voice radius statistic
VORDS => AAA:
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
Table 60 Output description
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
Voice dial program view
Default level
2: System level
Parameters
access-number: Access number (for example, 169 and 17909), a string of up to 31 characters consisting of
digits 0 through 9 and the wildcard ―.‖. The wildcard ―.‖ represents a digital character and must follow a digit
or appear separately.
all: Deletes all access numbers.
Examples
# Add the access number 17909 and enter access number view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909]
# Add the access number 179 and enter access number view.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 179..
[Sysname-voice-dial-anum179..]
# Delete the access number 17909.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] undo gw-access-number 17909
# Delete all access numbers.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] undo gw-access-number all
Delete all access numbers, are you sure? (Y/N) y
password-digit Description
Use password-digit to configure the number of digits in a password for some access number in the card
number/password process.
Use undo password-digit to restore the default number of digits in a password for some access number in the
card number/password process.
This command is unavailable for the caller number process with IVR. By default, the number of digits in a
password for some access number in the card number/password process is 6.
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.
Related commands: gw-access-number and process-config.
Syntax
password-digit password-digit
undo password-digit
View
Access number view
Default level
2: System level
Parameters
password-digit: Number of digits in a password, in the range of 1 to 16.
Examples
# Specify the number of digits in a password as 4 for the access number 17909.
<Sysname>system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] process-config cardnumber
[Sysname-voice-dial-anum17909] password-digit 4
# Restore the default number of digits in a password for the access number 17909.
[Sysname-voice-dial-anum17909] undo password-digit
process-config Description
Use process-config to specify a dialing process for an access number.
Use undo process-config to restore the default dialing process for an access number.
By default, the caller number process with IVR is specified for all access numbers.
Each access number has a specific dialing process. Calls originated by users who dial a certain access
number are established in accordance with the same dialing process.
The caller number process and the caller number process with IVR differ in two ways:
In the caller number process, after a user dials an access number, the voice gateway plays only dial
tones (long tones).
In the caller number process with IVR, after a user dials an access number, the voice gateway will play
prompt tones, requiring the user to dial a called number.
In the card number/password process, with the authentication function disabled, a user can enter any two
numbers as a card number and password respectively to make an IP call as long as they meet the length
requirements.
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.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] process-config cardnumber
# Restore the default dialing process for the access number 17909.
[Sysname-voice-dial-anum17909] undo process-config
redialtimes Description
Use redialtimes to configure the number of redial attempts in each dialing step for an access number.
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.
Related commands: gw-access-number and process-config.
Syntax
redialtimes redialtimes-number
undo redialtimes
View
Access number view
Default level
2: System level
Parameters
redialtimes-number: Number of redial attempts, in the range of 0 to 10. In the card number/password
process, this argument may refer to the times of reselecting a language or redialing a card number,
password, or a called number. In the caller number process with IVR, this argument may refer to the times of
reselecting a language or redialing a called number.
Examples
# Set the number of redial attempts to 4 for the access number 17909.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] process-config cardnumber
[Sysname-voice-dial-anum17909] redialtimes 4
reset voice radius statistic Description
Use reset voice radius statistic to clear statistics of messages exchanged between the voice RADIUS module,
CMC module, and AAA module.
Related commands: display voice radius statistic.
Syntax
reset voice radius statistic
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.
Related commands: gw-access-number and process-config.
Syntax
selectlanguage { enable | chinese | english }
undo selectlanguage
View
Access number view
Default level
2: System level
Parameters
enable: Enables the language selection function so that users can select a language to play prompt tones.
chinese: Plays prompt tones in Chinese.
english: Plays prompt tones in English.
Examples
# Configure the voice gateway to play prompt tones in English.
<Sysname> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] process-config voice-caller
[Sysname-voice-dial-anum17909] selectlanguage english
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> system-view
[Sysname] voice-setup
[Sysname-voice] dial-program
[Sysname-voice-dial] gw-access-number 17909
[Sysname-voice-dial-anum17909] timer two-stage dial-interval 5
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
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
Convention Description
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
Convention Description
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