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one200(voice-port)> ? analog-aoc-type - Analog AOC type (FXS port only) aoc-d-service - method for AOC-D behaviour aoc-e-service - method for AOC-E behaviour call-hold - Set VOIP call hold call-waiting - Set VOIP call waiting caller-id - Set VOIP caller id cas-conf - Configuration of cas signal analysis clock-source - Synchro source options (global over voice ports) coder-law - Set coder law dialing-timer - Set maximum time-out for receiving 1st digit (in sec) echo-cancellation - Set echo cancellation echo-cancellation-le - Set echo-cancellation-length echo-disable - For echo cancellation Modem:remove echo on 2100Hz phase reversal detection Voicemodem: modem + reactivate echo
when voice is back again end-of-dialing-timer - Digit timeout (in sec) to consider a call as complete exit - Exit intermediate mode force-clir - Set caller line identity request initial-ring - Initial ring tone in ms for caller-id...
... input-gain - Set input gain inter-digit - Set VOIP DTMF inter-digit duration (in sec) isdn-release-tone - set isdn-release-tone localy and force PI isdn-ringback-tone - set isdn-ringback-tone localy and force PI max-ringing - Maximum time for ringing before off_hook detection metering - metering choice no - no output-gain - Set output gain power-source-one - Set Power source 1 for all BRI voice-ports pulse-dial - Select country to validate pulse dial ring - Select country to define current ring shutdown - Shutdown for voice-port sig-conf - Configuration of signal analysis signal-analysis - Set signal transparency sntp-time - SNTP date/time inserted when ie is absent tone - Select a country to validate tone tone-level - Set level tone user-metering - User metering pulse profile user-ring - Modify the userdefined ring user-tone - Select the type of userdefined tone to modify: dial, network-failure, congestion, busy, callback without-loss-signal - Set without loss signal <cr>
one200(configure)>interface bri 5/0one200(config-if)> ? exit - exit isdn - Set isdn level no - no shutdown - shutdown
one200(configure)>interface pri 5/0one200(config-if)> ? exit - Exit intermediate mode framing - Set type of frames isdn - Set isdn level linecode - Select line physical code no - no physical-interface - Select the type of interface : E1 or T1 shutdown - Shutdown for the PRI interface
one200(isdn)> ? application-interfac - Set the application interface name exit - Exit to root node facility - facility message is transmit k-window - Set the value of k window layer1-emulation - Set the layer 1 emulation type layer2-emulation - Set the layer 2 emulation type life-line-hold - Life line hold for line 0 on ISDN Voice Board max - max modulo-window - Set the modulo window value n200-counter - Set the value of N200 counter n202-counter - Set the value of N202 counter no - no operator - Set the operator name protocol-emulation - Set the type of protocol emulation static-tei - Set the value of static tei t200-timer - Set the value of T200 timer... t310-timer - Set the value of T310 timer --> can be set to 100 for GSM calls tei-negotiation - Set the tei negociation mode <cr>one200(isdn)>
one200(isdn)> ? application-interfac - Set the application interface name exit - Exit to root node facility - message facility is transmit k-window - Set the value of k window layer2-emulation - Set the layer 2 emulation type max - max n200-counter - Set the value of N200 counter no - no operator - Set the operator name protocol-emulation - Set the type of protocol emulation t200-timer - Set the value of T200 timer t203-timer - Set the value of T203 timer t301-timer - Set the value of T301 timer t302-timer - Set the value of T302 timer t303-timer - Set the value of T303 timer t304-timer - Set the value of T304 timer t305-timer - Set the value of T305 timer t306-timer - Set the value of T306 timer t308-timer - Set the value of T308 timer t309-timer - Set the value of T309 timer t310-timer - Set the value of T310 timer t313-timer - Set the value of T313 timer
CLI(configure)# dial-peer voice pots <id>CLI(pots)# pots-group <id> CLI(pots)# port 5/<port>CLI(pots)# insert-calling-number <E164 number>CLI(pots)# no shutdownCLI(pots)# exit
Logical Local Voice Port
- For a outgoing Voip call, user part of From header field is based on 6C IE (calling party number) for BRI interface. For FXS port, that information must be added at the dial-peer voice pots adding “insert-calling-number”. That will be used also for 40x challenge on Invite method (to resolve user and its digest username and password.
one200(configure)> dial-peer voice pots 0one200(pots)> ? bearer-cap - Payload category direct-call - Set direct call number exit - Exit intermediate mode implicit-routing - Sets implicit routing to specified
pots group or voip dial peer insert-calling-numbe - Set VOIP insert calling number no - no port - Links local suscriber and voice port pots-group - Set VOIP pots group priority - Set priority service - to provide a service by the voice pots. shutdown - Shutdown for dial peer POTS suppress-calling-num - Set VOIP suppresion of the calling number <cr>one200(pots)>
Determine the SIP destination (User agent) for an outgoing call: Sig-protocol sip: Determines the signalling protocol Gw-ip-address: Determines the remote end point of the voice
call (SIP transaction messages and RTP/RTCP packets SIP end point may be another SIP gateway (UA) SIP end point may be a SIP phone or Softphone (UA) SIP end point may be a SIP proxy server (uses when different proxy
server have to be reached, voice-routing) This parameter is not required if prox-dns-add exists in sip-gateway.
CLI(configure)# dial-peer voice voip 0CLI(voip)# sig-protocol sipCLI(voip)# gw-ip-address <ip-address[:port]> <hostname>CLI(voip)# force-prackCLI(voip)# sip-sdp-on-alert {receive-only|send-receive|send-only}CLI(voip)# fax-relay {passthrough|t38|t38orpassthrough|t38nse} for t38orpassthrough priority {t38|passthrough}CLI(voip)# passthrough-mode {reinvite}CLI(voip)# dtmf-relay {in-band (rfc2833)|sip-info}CLI(voip)# no shutdown
Dial-peer VoIP (2/2)
force-prack # 100rel is added in supported header field. FromThere, proxy may require PRACK to ack a 1xx message.Sip-sdp-on-alert # About early media handling, requires to add SDP Message body at outgoing 180 message, requires to process early media for Incoming 180 and 183 messages.Fax-relay # Re-invite including T38 or G711 in SDP message bodypassthrough-mode reinvite # Require if voice call is establish for G729 and Fax-relay passthrough and/or modem-passthrough is validatedDtmf-relay # Transmission of dtmf digit to voip. OneOs doesn’tSupport incoming SIP INFO message (not useful)
one200(configure)> dial-peer voice voip 0one200(voip)> ? aoc-format - Set VOIP remote AOC coding format call-media-independa - Set VOIP call media independant dtmf-relay - Set VOIP dtmf relay exit - Exit from command node fast-connect - Set VOIP fast connect fax-relay - Set VOIP fax relay force-rec-inband - Force reception of inband in Alert gatekeeper - Set VOIP gatekeeper gw-ip-address - Set VOIP gateway h245-tunnel - Set VOIP H245 tunnel implicit-routing - Set implicit routing jitter - Set VOIP jitter jitter-compensation - Set VOIP jitter comp max-conn - Set VOIP maximum call allowed modem-passthrough - Set VOIP modem passthrough NdiInsourceAddress - Force NDI in H323 sourceAddress no - no shutdown - Shutdown voip dial peer silence-detection - Set VOIP silence detection t38-redundancy - Set VOIP T38 redundancy voip-coder-profile - Set VOIP coder profile
The following coders are supported: G.711 A law (64Kbps) G.711 law (64Kbps) G.729A (8 Kbps, no silence suppression) G.729AB (8 kbps, optional silence suppression)
CLI(configure)# sip-gateway bandwidth-control - Set SIP gateway - bandwidth control bridge-uri-host - Set bridge URI host characteristics bye-on-refer - Send bye when refer is received bye-on-refer-accept - Send bye when Refer Accept is received bye-timer - duration before bye message calling-number-checking - Check origin number is registered to process a call [default] callsig-port - Set SIP gateway - SIP listening port clip-privacy-uri - Define clip privacy predefined URI clip-unsubscribe-uri - Set predefined CLIP unsubscribe URI connect-timer - duration of waiting 200 OK device-host-name - Set sip gateway host name discard-3XX - Upon receiving a 3XX, the call is cleared exit - Exit from command node gw-interface - Output Interface category for SIP GW - default fastethernet 0 gw-interface-bw-ctrl - Set SIP gateway - gw interface bandwidth control invite-method-timeout - Invite methode Timeout. Timeout before receiving a final response invite-response-timer - duration of waiting first 1xx message max-bandwidth - Maximum Bandwidth allowed message-waiting-indication - We should attempt to receive message waiting indication no - no outbound-proxy - Set outbound proxy for all messages (Noted also SBC)
CLI(configure)# sip-gateway payload-64k-unrestricted - Payload for 64k unrestricted presentation-restricted - presentation restricted (Anonymous) prox-dns-add - Set Proxy characteristics prox-ka - Proxy keep alive value reg-dns-add - Set Registrar characteristics reg-failure-timer - Start when the UA SIP receives a 4xx, 5xx, 6xx response reg-interval-timeout - Set interval registration timeout reg-ka - Registrar keep alive value registration-timeout - Set registration timeout request-primitive-timer - Define Timeout for a Request SIP message shutdown - Shutdown SIP sig-dscp - DSCP field value for signalling packets sip-authentication - Set sip gateway username and password sip-called-number - get called number from sip invite message sip-uri-escape - escape # and * in sip URI sip-username - Set sip gateway ident softswitch-profile - softswitch type { default | broadworks } subscription-duration - Subscription duration value subscription-failed - Subscription failed duration value trunking-mode - set trunking mode uri-contact - Set type of URI in contact uri-from - Set type of URI in from user-agent - Do we include the user-agent header in the SIP INVITE message { include | exclude } voicemail-dns-add - Set VoiceMail characteristics
Translation of SIP username to E164 number: This feature is available for incoming call (dial-peer voip) A SIP user (SIP phone) may calls remote user using SIP URI
as [email protected] (called party) instead of its E164 number Voice-routing entry: sip-username converted to the prefix value
of this entry before to be sent to ISDN stack and populates the IE 70.
OneOs Rules to build the Request URI about Invite method: 1st: Utilisation of prox-dns-add value in the sip-gateway 2nd: If 1st doesn’t exist, utilisation of gw-ip-address value of
the dial-peer voice voip x of the corresponding voice route entry.
OneOs Rules to build the To header field for Invite method: 1st: utilisation of gw-ip-address value of the dial-
peer voice voip x of the corresponding voice route entry. 2nd: If 1st doesn’t exist, utilisation of prox-dns-add value in the
CLI# show voice voice-port bri index 0 voice port 5/0 protocol descriptor BRI_NT current state activated config state up layer 1 status activated attached vmoabri dial peer 0 number of voice communication 0 bri Tx frames on D channel 40 bri Rx frames on D channel 41
Outgoing calls : 102 Outgoing calls failures : 5 Physical Interface down : 0 Cause Class 0 (normal event) : 0 Cause Class 1 (normal event) : 5 Normal Cause (16) : 2 User busy (17) : 3 No answer (18) : 0 Cause Class 2 (unavailable ressources) : 0 Cause Class 3 (unavailable service) : 0 Cause Class 4 (service not provided) : 0 Cause Class 5 (invalid message) : 0 Cause Class 6 (protocol error) : 0 Cause Class 7 (interworking) : 0
CLI# show voice voice-port bri index 0 voice port 5/0 protocol descriptor BRI_NT current state activated config state up layer 1 status activated attached vmoabri dial peer 0 number of voice communication 0 bri Tx frames on D channel 40 bri Rx frames on D channel 41
Outgoing calls : 102 Outgoing calls failures : 5 Physical Interface down : 0 Cause Class 0 (normal event) : 0 Cause Class 1 (normal event) : 5 Normal Cause (16) : 2 User busy (17) : 3 No answer (18) : 0 Cause Class 2 (unavailable ressources) : 0 Cause Class 3 (unavailable service) : 0 Cause Class 4 (service not provided) : 0 Cause Class 5 (invalid message) : 0 Cause Class 6 (protocol error) : 0 Cause Class 7 (interworking) : 0
CLI# show voice voice-port pri index 0 voice port 5/0 physical type E1 protocol descriptor E1_PRI current state activated config state up layer 1 status deactivated number of voice communications 0 pri AIS occurence 0 pri RDI occurence 0
Outgoing calls : 67 Outgoing calls failures : 3 Physical Interface down : 0 Cause Class 0 (normal event) : 0 Cause Class 1 (normal event) : 3 Normal Cause (16) : 0 User busy (17) : 3 No answer (18) : 0 Cause Class 2 (unavailable ressources) : 0 Cause Class 3 (unavailable service) : 0 Cause Class 4 (service not provided) : 0 Cause Class 5 (invalid message) : 0 Cause Class 6 (protocol error) : 0 Cause Class 7 (interworking) : 0
CLI# show voice voice-port pri index 0 voice port 5/0 physical type E1 protocol descriptor E1_PRI current state activated config state up layer 1 status deactivated number of voice communications 0 pri AIS occurence 0 pri RDI occurence 0
Outgoing calls : 67 Outgoing calls failures : 3 Physical Interface down : 0 Cause Class 0 (normal event) : 0 Cause Class 1 (normal event) : 3 Normal Cause (16) : 0 User busy (17) : 3 No answer (18) : 0 Cause Class 2 (unavailable ressources) : 0 Cause Class 3 (unavailable service) : 0 Cause Class 4 (service not provided) : 0 Cause Class 5 (invalid message) : 0 Cause Class 6 (protocol error) : 0 Cause Class 7 (interworking) : 0
one200> show voice dial-peer voice voip type index <port id> [reset]orone200> show voice dial-peer voice voip type global[reset]
type may be : -current : statistics on current calls-outgoing : outgoing calls only-incoming : incoming calls only-user-plan : voice & fax only-all (default) : all the statistics are provided
one200> show voice dial-peer voice voip type index <port id> [reset]orone200> show voice dial-peer voice voip type global[reset]
type may be : -current : statistics on current calls-outgoing : outgoing calls only-incoming : incoming calls only-user-plan : voice & fax only-all (default) : all the statistics are provided
RTP statistics Number of transmitted packets 1237 Number of received packets 1234 Number of transmitted bytes 101484 Number of received bytes 101098 Number of excessive jitter events 3 Number of lost packets 0 Number of invalid packets 0 Number of calls with frame error rate total <0.01<0.1<0.5<1<5>=5 3 0 0 2 1 0 0 Modem passthrough Number of switching to modem mode 0 T38 FAX Calls Number of outgoing fax 0 Number of incoming fax 0 Number of failures 0 Request Mode failure 0 Pre-message procedure failure 0 Page failure 0 Number of transmitted packets 0 Number of received packets 0 Number of transmitted bytes 0 Number of received bytes 0 Number of lost packets 0
vxTarget>event filter - Add/remove events filters manager - Add a SNMP manager no - No recover - Recover events from memoryvxTarget>event filter add - Add an event filter remove - Remove a events filter from the tablevxTarget>event filter add vox ALL - All families from vox group GEN - GEN VOATM - VOATM VOIP - VOIPvxTarget>event filter add vox voip <subfam> - <ALL | ControlPlan | UserPlan> <fam2> - <GEN | VOATM>vxTarget>event filter add vox voip all show
vxTarget>event filter - Add/remove events filters manager - Add a SNMP manager no - No recover - Recover events from memoryvxTarget>event filter add - Add an event filter remove - Remove a events filter from the tablevxTarget>event filter add vox ALL - All families from vox group GEN - GEN VOATM - VOATM VOIP - VOIPvxTarget>event filter add vox voip <subfam> - <ALL | ControlPlan | UserPlan> <fam2> - <GEN | VOATM>vxTarget>event filter add vox voip all show
RTP sessions history Gives complete statistics about the 200 last RTP sessions
CLI> show voice rtpcall full any ind 2 2 - 01/04/01 00h47m24s RTP 192.168.1.1:16384 – 192.168.1.111:16386 Play time (voice) : 00h00m46s Tx Coder : G729 / 20 ms ; Rx Coder : G729 VAD enabled local / remote : no / no ERL : 15 dB ACOM : 32 dB RTP Packets received (DSP / Uplink) : 2337 / 2337 lost : 0 out of sequence : 0 invalid : 0 RTP Packets transmitted (DSP / Uplink) : 2338 / 2338 lost (RTCP reported) : 0 Jitter parameter : 100 ms Number of Excessive Jitter events : 1
CLI> show voice rtpcall full any ind 2 2 - 01/04/01 00h47m24s RTP 192.168.1.1:16384 – 192.168.1.111:16386 Play time (voice) : 00h00m46s Tx Coder : G729 / 20 ms ; Rx Coder : G729 VAD enabled local / remote : no / no ERL : 15 dB ACOM : 32 dB RTP Packets received (DSP / Uplink) : 2337 / 2337 lost : 0 out of sequence : 0 invalid : 0 RTP Packets transmitted (DSP / Uplink) : 2338 / 2338 lost (RTCP reported) : 0 Jitter parameter : 100 ms Number of Excessive Jitter events : 1
One_training>auto-call <called> - called number: up to 21 characters <0..9, #, *>One_training>auto-call 0141877422 <calling> - calling number: up to 21 characters <0..9, #, *> <pots-number> - pots: 0..29 <bearer> - bearer capability < voice | data | voiceband > overlap - units in milliseconds: 0..2000 <0 means no overlap used> <cr>One_training>auto-call 0141877422 240888200517:50:17.677 Info vox factory test 1 call-id: 4, ident: auto-call, CALL IN PROGRESS Calling=2408882005 Called=0141877422.one100_interopBW>17:50:17.678 Info vox voip controlplan 3 Incoming call on local pots: 0, calling:2408882005, called: 0141877422, call-id: 4.17:50:17.710 Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 2408882005, called:0141877422, call-id: 4.17:50:27.660 Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED cause=no codec.17:50:27.661 Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED on pots cause=[Normal call clearing].