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Cisco IOS Voice Command Reference - S commands
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sccp through service-type call-check
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session through sgcp tse payload
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show aal2 profile through show call filter match-list
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show call history fax through show debug condition
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show dial-peer through show gatekeeper zone prefix
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show gateway through show modem relay statistics
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show mrcp client session active through show sip dhcp
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show sip service through show trunk hdlc
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show vdev through show voice statistics memory-usage
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show voice trace through shutdown (voice-port)
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signal through srv version
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ss7 mtp2-variant through switchover method
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Index
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Feedback
Contents
show voice trace through shutdown (voice-port)
- show voice trace
- show voice translation-profile
- show voice translation-rule
- show voice trunk-conditioning signaling
- show voice trunk-conditioning supervisory
- show voice vtsp
- show voip debug version
- show voip fpi call-rate
- show voip fpi calls
- show voip fpi stats
- show voip htsp
- show voip recmsp session
- show voip rtp connections
- show voip rtp forking
- show voip trunk group
- show vrm active_calls
- show vrm vdevices
- show vsp
- show wsapi
- show xcsp port
- show xcsp slot
- shut
- shutdown (Annex G neighbor)
- shutdown (Annex G)
- shutdown (dial-peer)
- shutdown (DSP Farm profile)
- shutdown (gatekeeper)
- shutdown (gateway)
- shutdown (mediacard)
- shutdown (auto-config application)
- shutdown (RLM)
- shutdown (settlement)
- shutdown (voice-port)
show voice trace
To display the call trace information about a specified port, use the show voice trace command in privileged EXEC mode.
Usage Guidelines
Use the show voice tracecommandto display the call trace information about specified port. The field descriptions are self-explanatory.
Examples
The following is sample output from the show voice tracecommand:
Router# show voice trace 1/1/1 detail 1/1/1 Stack 0: State Transitions: timestamp (state, event) -> (state, event) ... 96.732 (S_OPEN_PEND, E_DSP_INTERFACE_INFO) -> 96.732 (S_DOWN, E_HTSP_IF_INSERVICE) -> 97.092 (S_OPEN_PEND, E_HTSP_GO_UP) -> Event Counts (zeros not shown): (event, count) (E_HTSP_IF_INSERVICE, 1) :(E_HTSP_GO_UP, 1) :(E_DSP_INTERFACE_INFO, 1) : State Counts (zeros not shown): (state, count) (S_OPEN_PEND, 2) :(S_DOWN, 1) : Stack 1: State Transitions: timestamp (state, event) -> (state, event) ... 97.092 (DID_NULL, E_DSP_SIG_0100) -> 97.092 (DID_INIT, E_HTSP_INSERVE) -> 97.092 (DID_PENDING, E_DSP_SIG_0100) -> Event Counts (zeros not shown): (event, count) (E_HTSP_INIT, 1) :(E_HTSP_INSERVE, 1) :(E_DSP_SIG_0100, 2) : State Counts (zeros not shown): (state, count) (DID_NULL, 2) :(DID_INIT, 1) :(DID_PENDING, 1) :show voice translation-profile
To display one or more translation profiles, use the show voice translation-profile command in privileged EXEC mode.
Examples
The following sample output displays all the voice translation profiles in ascending order:
Router# show voice translation-profile sort ascending Translation Profile: 1 Rule for Calling number: Rule for Called number: 1 Rule for Redirect number: Translation Profile: 2 Rule for Calling number:1 Rule for Called number: 2 Rule for Redirect number: Translation Profile: 6 Rule for Calling number:1 Rule for Called number: 6 Rule for Redirect number:2The table below describes the fields shown in this output.
Table 1 show voice translation-profile Field Descriptions Field
Description
Translation Profile
Name of the translation profile.
Rule for Called number
Number of the rule used for translating called numbers. If the field is blank, this translation profile does not have a rule assigned to that number type.
Rule for Calling number
Number of the rule used for translating calling numbers. If the field is blank, this translation profile does not have a rule assigned to that number type.
Rule for Redirect number
Number of the rule used for translating redirect numbers. If the field is blank, this translation profile does not have a rule assigned to that number type.
show voice translation-rule
To display one or more translation rules, use the show voice translation-rule command in privileged EXEC mode.
Examples
The following sample output displays the translation rule number 6:
Router# show voice translation-rule 6 Translation-rule tag: 6 Rule 1: Match pattern: 65088801.. Replace pattern: 6508880101 Match type: none Replace type: none Match plan: none Replace plan: noneThe following sample output displays all the translation rules in ascending order:
Router# show voice translation-rule sort ascending Translation-rule tag: 1 Rule 3: Match pattern: 5108880... Replace pattern: 5108880101 Match type: none Replace type: none Match plan: none Replace plan: none Rule 4: Match pattern: 510890.... Replace pattern: 5108880101 Match type: none Replace type: none Match plan: none Replace plan: none Translation-rule tag: 2 Rule 1: Match pattern: 51088802.. Replace pattern: 5108880101 Match type: none Replace type: none Match plan: none Replace plan: none Rule 2: Match pattern: 51088803.. Replace pattern: 5108880101 Match type: none Replace type: none Match plan: none Replace plan: none Rule 3: Match pattern: 510889.... Replace pattern: 5108880101 Match type: none Replace type: none Match plan: none Replace plan: none Rule 4: Match pattern: 510890.... Replace pattern: 5108880101 Match type: none Replace type: none Match plan: none Replace plan: noneThe table below describes the fields shown in this output.
Table 2 show voice translation-rule Field Descriptions Field
Description
Translation-rule tag
Number of the translation rule.
Rule
Number of the rule defined within the translation rule.
Match pattern
SED-like expression used to match incoming call information.
Replace pattern
SED-like expression used to replace match-pattern in the call information.
Match type
Type of incoming calls to match.
Replace type
Type to replace Match type.
Match plan
Plan of incoming calls to match.
Replace plan
Plan to replace Match plan.
Related Commands
Command
Description
rule (voice translation-rule)
Defines the SED expressions for translating calls.
test voice translation-rule
Tests the rules in a translation-rule definition.
voice translation-rule
Initiates a voice translation-rule definition.
voice translation-profile
Initiates a voice translation-profile definition.
show voice trunk-conditioning signaling
To display the status of trunk-conditioning signaling and timing parameters for a voice port, use the show voice trunk-conditioning signaling command in user EXEC or privileged EXEC mode.
Command History
Release
Modification
12.0(3)XG
This command was introduced on the Cisco MC3810 as the show voice permanent-call command.
12.0(4)T
This command was integrated into Cisco IOS Release 12.0(4)T.
12.0(7)XK
This command was renamed show voice trunk-conditioning signaling.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)T
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
Usage Guidelines
This command displays the trunk signaling status for analog and digital voice ports on the Cisco 2600 series and the Cisco 3600 series routers.
Examples
The following is sample output from the show voice trunk-conditioning signaling summary command:
Router# show voice trunk-conditioning signaling summary 2/0/0 is shutdown 2/0/1 is shutdown 3/0:0 8 is shutdown 3/0:1 1 is shutdown 3/0:2 2 is shutdown 3/0:3 3 is shutdown 3/0:5 5 is shutdown 3/0:6(6) : status : 3/0:7 7 is shutdown 3/1:0 8 is shutdown 3/1:1 1 is shutdown 3/1:3 3 is shutdown 3/1:5 5 is shutdown 3/1:7 7 is shutdownThe following is sample output from the show voice trunk-conditioning signaling command for voice port 3/0:6:
Router# show voice trunk-conditioning signaling 3/0:6 hardware-state ACTIVE signal type is NorthamericanCAS status : forced playout pattern = STOPPED trunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0The table below describes significant fields in these outputs.
Related Commands
Command
Description
show dial-peer voice
Displays the configuration for all VoIP and POTS dial peers configured on the router.
show voice dsp
Shows the current status of all DSP voice channels.
show voice port
Displays configuration information about a specific voice port.
show voice trunk-conditioning supervisory
Displays the status of trunk supervision and configuration parameters for voice ports.
show voice trunk-conditioning supervisory
To display the status of trunk supervision and configuration parameters for a voice port, use the show voice trunk-conditioning supervisory command in user EXEC or privileged EXEC mode.
Command History
Release
Modification
12.0(7)XK
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810 platforms.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.1(3)T
This command was implemented on the Cisco 2600 series and Cisco 3600 series.
12.4(15)T10
The output of thiscommand was modified to report values configured by the signal timing idle suppress-voice command. The values for the suppress-voice and resume-voice keywords are shown as the "idle = seconds" and "idle_off = milliseconds" fields, respectively.
Usage Guidelines
This command displays the trunk supervision and configuration status for analog and digital voice ports.
Examples
The following is sample output from the show voice trunk-conditioning supervisory summary command for all voice ports:
Router# show voice trunk-conditioning supervisory summary 2/0/0 is shutdown 2/0/1 is shutdown 3/0:0 8 is shutdown 3/0:1 1 is shutdown 3/0:2 2 is shutdown 3/0:3 3 is shutdown 3/0:5 5 is shutdown 3/0:6(6) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/0:7(7) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/1:0(8) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/1:1(1) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/1:3(3) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/1:5(5) is shutdown 3/1:7(7) is shutdownThe following is sample output from the show voice trunk-conditioning supervisory command for voice port 3/0:6:
Router# show voice trunk-conditioning supervisory 3/0:6 3/0:6(6) : state : TRUNK_SC_CONNECT, voice : on, signal : on, master status: trunk connected sequence oos : idle and oos pattern :rx_idle = 0x0 rx_oos = 0xF timing : idle = 0, restart = 0, standby = 0, timeout = 40 supp_all = 0, supp_voice = 0, keep_alive = 5 timer: oos_ais_timer = 0, timer = 0The following shows a sample trunk conditioning setting for the voice class permanent command and sample output from the s how voice trunk-conditioning supervisory command that shows the values for the timeout timing field:
! voice class permanent 1 signal pattern idle transmit 0101 signal pattern idle receive 0101 signal pattern oos transmit 1111 signal pattern oos receive 0101 signal timing idle suppress-voice 10 resume-voice 150 ! Router# show voice trunk-conditioning supervisory SLOW SCAN 0/0/0:0(1) : state : TRUNK_SC_CONNECT, voice : off , signal : on ,slave status: rcv IDLE, trunk connected sequence oos : idle and oos pattern :rx_idle = 0101 rx_oos = 0101 tx_idle = 0101 tx_oos = 1111 timeout timing : idle = 10, idle_off = 150, restart = 0, standby = 0, timeout = 30 supp_all = 0, supp_voice = 0, keep_alive = 5 timer: oos_ais_timer = 0, timer = 0The table below describes the significant fields shown in the display.
Table 4 show voice trunk-conditioning supervisory Field Descriptions Field
Description
idle
Timer setting (in seconds) configured by the suppress-voice option of the signal timing idle suppress-voice command.
idle_off
Timer setting (in milliseconds) configured by the resume-voice option of the signal timing idle suppress-voice command.
keep_alive
Signaling packets periodically sent to the far end, even if there is no signal change. These signaling packets function as keep alive messages.
master
Voice port configured as "connect trunk xxxx ."
oos_ais_timer
Time since the signaling packet with alarm indication signal (AIS) indicator was received.
pattern
4-bit signaling pattern.
restart
Restart timeout after far end is out-of-service (OOS).
rx-idle
Signaling bit pattern indicating that the far end is idle.
rx-oos
Signaling bit pattern sent to the PBX indicating that the network is OOS.
standby
Time before the slave side goes back to standby after the far end goes OOS.
supp_all
Timeout before suppressing transmission of voice and signaling packets to the far end after detection of PBX OOS.
supp_voice
Timeout before suppressing transmission of voice packet to the far end after detection of PBX OOS.
timeout
Timeout for nonreceipt of keepalive packets before the far end is considered to be OOS.
timeout timing
Delay between the detection of incoming seizure and when the digital signal processor (DSP)-to-Cisco IOS interaction to open up the audio path is initiated.
TRUNK_SC_CONNECT
Trunk conditioning supervisory component status.
Related Commands
Command
Description
show dial-peer voice
Displays the configuration for all VoIP and POTS dial peers configured on the router.
show voice dsp
Displays the current status of all DSP voice channels.
show voice port
Displays configuration information about a specific voice port.
show voice trunk-conditioning signaling
Displays the status of trunk-conditioning signaling and timing parameters for a voice port.
voice-class permanent
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a voice port.
show voice vtsp
To display information about the voice port configuration and Voice Telephony Service Provider (VTSP), use the show voice vtsp command in privileged EXEC mode.
Syntax Description
call
Displays the call control block information.
dspstats
(Optional) Displays the selective statistics of digital signal processor (DSP) voice channels.
fsm
(Optional) Displays information about the Finite State Machine Dump (FSM).
log call-ID
(Optional) Displays the call related logs. If a call ID is specified, this command displays the status of a specific call. The call ID value range is from 1 to 4294967295
verbose
(Optional) Displays the verbose output.
fork
Displays the media forking information.
dsp-status
Displays the status of media forking in the DSP.
call-ID
(Optional) Displays the status of the call. The value range is from 0x0 to 0xFFFFFFFF. >
Usage Guidelines
Use the show voice vtsp command to display information about the voice port configuration.
Examples
The following is sample output from the show voice vtsp command:
Router# show voice vtsp call dspstats 0x833 ***DSP VOICE TX STATISTICS*** Tx Vox/Fax Pkts: 1337, Tx Sig Pkts: 0, Tx Comfort Pkts: 181 Tx Dur(ms): 46840, Tx Vox Dur(ms): 26740, Tx Fax Dur(ms): 0 ***DSP VOICE RX STATISTICS*** Rx Vox/Fax Pkts: 1347, Rx Signal Pkts: 0, Rx Comfort Pkts: 180 Rx Dur(ms): 46840, Rx Vox Dur(ms): 23300, Rx Fax Dur(ms): 0 Rx Non-seq Pkts: 0, Rx Bad Hdr Pkts: 0 Rx Early Pkts: 0, Rx Late Pkts: 0 ***DSP VOICE VP_DELAY STATISTICS*** Clk Offset(ms): 80, Rx Delay Est(ms): 50 Rx Delay Lo Water Mark(ms): 50, Rx Delay Hi Water Mark(ms): 70 ***DSP VOICE VP_ERROR STATISTICS*** Predict Conceal(ms): 0, Interpolate Conceal(ms): 0 Silence Conceal(ms): 0, Retroact Mem Update(ms): 0 Buf Overflow Discard(ms): 0, Talkspurt Endpoint Detect Err: 0 ***DSP LEVELS*** TDM Bus Levels(dBm0): Rx -68.5 from PBX/Phone, Tx -4.4 to PBX/Phone TDM ACOM Levels(dBm0): +64.1, TDM ERL Level(dBm0): +10.0 TDM Bgd Levels(dBm0): -80.0, with activity being silence ***DSP VOICE ERROR STATISTICS*** Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0 ***DSP VOICE GSMAMR-NB STATISTICS*** EncodingRate: 7 DecodingRate: 7 numEncodeChanges: 0 numDecodeChanges: 0 numCRCFail: 0 numFrameBadQuality: 0 numInvalidCMR: 0 numInvalidFrameType: 0show voip debug version
To display the current version of the Voice over IP debug structure, use the show voip debug versioncommand in privileged EXEC mode.
show voip fpi call-rate
To display the average call rates at the forwarding plane interface, use the show voip fpi call-rate command in privileged EXEC mode.
Syntax Description
interval Displays the message rates at the FPI interface
seconds The number of seconds for the interval. The range is from 1 to 300
history Specifies how far back information is kept and displayed.
seconds The number of seconds that will be displayed. The range is from 1 to 86400.
Command History
Usage Guidelines
This command displays the call-rate data that is collected on the forwarding plane interface when the debug voip fpi call-rate is enabled .
Examples
The following shows the output for the show voip fpi call-rate command
Router# show voip fpi call-rate interval 1 history 1 ------ ------ ------ ------ ------ ------ ------ Sec ADD MOD DEL EVT_UP EVT_DN CPU 5S ------ ------ ------ ------ ------ ------ ------ 67 0 0 0 0 0 0 ------ ------ ------ ------ ------ ------ ------show voip fpi calls
To display call information for TDM and IVR calls in the Forwarding Plane Interface (FPI), use the show voip fpi calls command in privileged EXEC mode.
Syntax Description
all (Optional) Displays the detailed statistics for all calls in the FPI where the collection processes have been enabled.
confID identifier (Optional) Displays detailed call information for a call based on the conference ID.
callID identifier (Optional) Displays detailed call information for a call based on the call ID.
correlator identifier (Optional) Displays detailed call information for a call based on the correlator ID.
Command History
Examples
The following are sample output from the show voip fpi calls command
Router# show voip fpi calls Number of Calls : 2 ---------- ---------- ---------- ----------- --------------- --------------- confID correlator AcallID BcallID state event ---------- ---------- ---------- ----------- --------------- --------------- 20 20 87 88 ALLOCATED DETAIL_STAT_RSP 21 21 89 90 ALLOCATED DETAIL_STAT_RSPRouter# show voip fpi calls confID 20 --------------------------------------------------------------------------- VoIP-FPI call entry details: --------------------------------------------------------------------------- Call Type : IP_IP confID : 20 correlator : 20 call_state : ALLOCATED last_event : DETAIL_STAT_RSP alloc_start_time : 2737426765 modify_start_time: 0 delete_start_time: 0 Media Type(SideA): RTP Media Type(SideB): RTP --------------------------------------------------------------------------- FPI State Machine Stats: ------------------------ create_req_call_entry_inserted : 1 call_create_req_fsm_successful : 1 call_provision_rsp_ok : 1 call_provision_rsp_fsm_successful : 1 event_ind_media_up_to_app : 2 --------------------------------------------------------------------------------------------- SIDE_A RTP details - gccb=0x7FE69FA11C08 --------------------------------------------------------------------------------------------- confID : 20 fpi_user_data : 20 callID : 87 dstCallID : 88 mainstcallID : 87 srcport : 16552 dstport : 16580 DP add_sent : 1 dp_add_fail : 0 dp_add_pending : 0 dp_delete_sent : 0 dp_delete_waiting: 0 dp_delete_done : 0 final_stats_pend : 0 ha_create_sent : 1 is_video : 0 media_type : 0 is dspfarm xcode : No is conference : No stream_type : VOICE rtp_type : SENDRECV --------------------------------------------------------------------------------------------- SIDE_B RTP details - gccb=0x7FE6A9B5A960 --------------------------------------------------------------------------------------------- confID : 20 fpi_user_data : 20 callID : 88 dstCallID : 87 mainstcallID : 88 srcport : 16554 dstport : 16400 DP add_sent : 1 dp_add_fail : 0 dp_add_pending : 0 dp_delete_sent : 0 dp_delete_waiting: 0 dp_delete_done : 0 final_stats_pend : 0 ha_create_sent : 1 is_video : 0 media_type : 0 is dspfarm xcode : No is conference : No stream_type : VOICE rtp_type : SENDRECV --------------------------------------------------------------------------------------------- Detailed Stats from DataPlane: ------------------------------ mgm_handle : 20 ------------------------------------------------------------------- Call Present in : FMAN RP FMAN FP CPP ---------------------------------------- YES YES YES ------------------------------------------------------------------- Field sideA sideB ------------------------------------------------------------------- dtmf_payload_type 0 0 redundant_data_pyld_type 255 255 tos_mask 0 0 dtmf_flags 0 0 ucode_flags 5 5 local_port 16552 16554 remote_port_tx 16580 16400 remote_port_rx 16580 16400 session_id 0x30000050 0x30000052 hairpin_prtnr_null(ucode) NULL NULL hairpin_prtnr_callid 0 0 dsp_interface_null NULL NULL ------------------------------------------------------------------- DSP Resource Used : NoRouter# show voip fpi calls callid 87 --------------------------------------------------------------------------- VoIP-FPI call entry details: --------------------------------------------------------------------------- Call Type : IP_IP confID : 20 correlator : 20 call_state : ALLOCATED last_event : DETAIL_STAT_RSP alloc_start_time : 2737426765 modify_start_time: 0 delete_start_time: 0 Media Type(SideA): RTP Media Type(SideB): RTP --------------------------------------------------------------------------- FPI State Machine Stats: ------------------------ create_req_call_entry_inserted : 1 call_create_req_fsm_successful : 1 call_provision_rsp_ok : 1 call_provision_rsp_fsm_successful : 1 event_ind_media_up_to_app : 2 --------------------------------------------------------------------------------------------- SIDE_A RTP details - gccb=0x7FE69FA11C08 --------------------------------------------------------------------------------------------- confID : 20 fpi_user_data : 20 callID : 87 dstCallID : 88 mainstcallID : 87 srcport : 16552 dstport : 16580 DP add_sent : 1 dp_add_fail : 0 dp_add_pending : 0 dp_delete_sent : 0 dp_delete_waiting: 0 dp_delete_done : 0 final_stats_pend : 0 ha_create_sent : 1 is_video : 0 media_type : 0 is dspfarm xcode : No is conference : No stream_type : VOICE rtp_type : SENDRECV --------------------------------------------------------------------------------------------- SIDE_B RTP details - gccb=0x7FE6A9B5A960 --------------------------------------------------------------------------------------------- confID : 20 fpi_user_data : 20 callID : 88 dstCallID : 87 mainstcallID : 88 srcport : 16554 dstport : 16400 DP add_sent : 1 dp_add_fail : 0 dp_add_pending : 0 dp_delete_sent : 0 dp_delete_waiting: 0 dp_delete_done : 0 final_stats_pend : 0 ha_create_sent : 1 is_video : 0 media_type : 0 is dspfarm xcode : No is conference : No stream_type : VOICE rtp_type : SENDRECV --------------------------------------------------------------------------------------------- Detailed Stats from DataPlane: ------------------------------ mgm_handle : 20 ------------------------------------------------------------------- Call Present in : FMAN RP FMAN FP CPP ---------------------------------------- YES YES YES ------------------------------------------------------------------- Field sideA sideB ------------------------------------------------------------------- dtmf_payload_type 0 0 redundant_data_pyld_type 255 255 tos_mask 0 0 dtmf_flags 0 0 ucode_flags 5 5 local_port 16552 16554 remote_port_tx 16580 16400 remote_port_rx 16580 16400 session_id 0x30000050 0x30000052 hairpin_prtnr_null(ucode) NULL NULL hairpin_prtnr_callid 0 0 dsp_interface_null NULL NULL ------------------------------------------------------------------- DSP Resource Used : NoRouter# show voip fpi calls all Number of Calls : 2 --------------------------------------------------------------------------- VoIP-FPI call entry details: --------------------------------------------------------------------------- Call Type : IP_IP confID : 24 correlator : 24 call_state : ALLOCATED last_event : DETAIL_STAT_RSP alloc_start_time : 2902404766 modify_start_time: 0 delete_start_time: 0 Media Type(SideA): RTP Media Type(SideB): RTP --------------------------------------------------------------------------- FPI State Machine Stats: ------------------------ create_req_call_entry_inserted : 1 call_create_req_fsm_successful : 1 call_provision_rsp_ok : 1 call_provision_rsp_fsm_successful : 1 event_ind_media_up_to_app : 2 --------------------------------------------------------------------------------------------- SIDE_A RTP details - gccb=0x7FE69FA11C08 --------------------------------------------------------------------------------------------- confID : 24 fpi_user_data : 24 callID : 95 dstCallID : 96 mainstcallID : 95 srcport : 16568 dstport : 16580 DP add_sent : 1 dp_add_fail : 0 dp_add_pending : 0 dp_delete_sent : 0 dp_delete_waiting: 0 dp_delete_done : 0 final_stats_pend : 0 ha_create_sent : 1 is_video : 0 media_type : 0 is dspfarm xcode : No is conference : No stream_type : VOICE rtp_type : SENDRECV --------------------------------------------------------------------------------------------- SIDE_B RTP details - gccb=0x7FE6A9B5A960 --------------------------------------------------------------------------------------------- confID : 24 fpi_user_data : 24 callID : 96 dstCallID : 95 mainstcallID : 96 srcport : 16570 dstport : 16400 DP add_sent : 1 dp_add_fail : 0 dp_add_pending : 0 dp_delete_sent : 0 dp_delete_waiting: 0 dp_delete_done : 0 final_stats_pend : 0 ha_create_sent : 1 is_video : 0 media_type : 0 is dspfarm xcode : No is conference : No stream_type : VOICE rtp_type : SENDRECV --------------------------------------------------------------------------------------------- Detailed Stats from DataPlane: ------------------------------ mgm_handle : 24 ------------------------------------------------------------------- Call Present in : FMAN RP FMAN FP CPP ---------------------------------------- YES YES YES ------------------------------------------------------------------- Field sideA sideB ------------------------------------------------------------------- dtmf_payload_type 0 0 redundant_data_pyld_type 255 255 tos_mask 0 0 dtmf_flags 0 0 ucode_flags 5 5 local_port 16568 16570 remote_port_tx 16580 16400 remote_port_rx 16580 16400 session_id 0x30000060 0x30000062 hairpin_prtnr_null(ucode) NULL NULL hairpin_prtnr_callid 0 0 dsp_interface_null NULL NULL ------------------------------------------------------------------- DSP Resource Used : No ------------------------------------------------------------ --------------------------------------------------------------------------- VoIP-FPI call entry details: --------------------------------------------------------------------------- Call Type : IP_IP confID : 25 correlator : 25 call_state : ALLOCATED last_event : DETAIL_STAT_RSP alloc_start_time : 2902505765 modify_start_time: 0 delete_start_time: 0 Media Type(SideA): RTP Media Type(SideB): RTP --------------------------------------------------------------------------- FPI State Machine Stats: ------------------------ create_req_call_entry_inserted : 1 call_create_req_fsm_successful : 1 call_provision_rsp_ok : 1 call_provision_rsp_fsm_successful : 1 event_ind_media_up_to_app : 1 --------------------------------------------------------------------------------------------- SIDE_A RTP details - gccb=0x7FE6A9B9CFA8 --------------------------------------------------------------------------------------------- confID : 25 fpi_user_data : 25 callID : 97 dstCallID : 98 mainstcallID : 97 srcport : 16572 dstport : 16584 DP add_sent : 1 dp_add_fail : 0 dp_add_pending : 0 dp_delete_sent : 0 dp_delete_waiting: 0 dp_delete_done : 0 final_stats_pend : 0 ha_create_sent : 0 is_video : 0 media_type : 0 is dspfarm xcode : No is conference : No stream_type : VOICE rtp_type : SENDRECV --------------------------------------------------------------------------------------------- SIDE_B RTP details - gccb=0x7FE69FA132F8 --------------------------------------------------------------------------------------------- confID : 25 fpi_user_data : 25 callID : 98 dstCallID : 97 mainstcallID : 98 srcport : 16574 dstport : 16404 DP add_sent : 1 dp_add_fail : 0 dp_add_pending : 0 dp_delete_sent : 0 dp_delete_waiting: 0 dp_delete_done : 0 final_stats_pend : 0 ha_create_sent : 1 is_video : 0 media_type : 0 is dspfarm xcode : No is conference : No stream_type : VOICE rtp_type : SENDRECV --------------------------------------------------------------------------------------------- Detailed Stats from DataPlane: ------------------------------ mgm_handle : 25 ------------------------------------------------------------------- Call Present in : FMAN RP FMAN FP CPP ---------------------------------------- YES YES YES ------------------------------------------------------------------- Field sideA sideB ------------------------------------------------------------------- dtmf_payload_type 0 0 redundant_data_pyld_type 255 255 tos_mask 0 0 dtmf_flags 0 0 ucode_flags 0 5 local_port 16572 16574 remote_port_tx 16584 16404 remote_port_rx 16584 16404 session_id 0x30000064 0x30000066 hairpin_prtnr_null(ucode) NULL NULL hairpin_prtnr_callid 0 0 dsp_interface_null NULL NULL ------------------------------------------------------------------- DSP Resource Used : Noshow voip fpi stats
To display the TDM and IVR statistics and error counters in the Forwarding Plane Interface (FPI), use the show voip fpi stats command in privileged EXEC mode.
Syntax Description
Command History
Examples
The following is a sample output from the show voip fpi stats command:
Router# show voip fpi stats *********************** ACTIVE ******************************* IDLE ALLOCATING ALLOCATED MODIFYING CREATE_REQ 25 0 0 0 MODIFY_REQ 0 0 21 17 DELETE_REQ 0 0 22 3 GET_STATS_REQ 0 0 0 0 PROV_RSP_OK 0 25 0 18 PROV_RSP_FAIL 0 0 0 0 DELETE_RSP 0 0 0 0 GET_STATS_RSP 0 0 0 0 STATS_TMR_EXP 0 0 0 0 TMR_EXPIRY 0 0 0 0 CREATE_STRM_REQ 0 0 0 0 MODIFY_STRM_REQ 0 0 0 0 DELETE_STRM_REQ 0 0 0 0 DETAIL_STAT_REQ 0 0 631 0 DETAIL_STAT_RSP 0 0 631 0 DT_STAT_TMR_EXP 0 0 0 0 DELETING ALLOC_MOD_PEND MODIFY_MOD_PEND DELETE_PENDING CREATE_REQ 0 0 0 0 MODIFY_REQ 0 0 8 0 DELETE_REQ 0 0 0 0 GET_STATS_REQ 0 0 0 0 PROV_RSP_OK 0 0 17 3 PROV_RSP_FAIL 0 0 0 0 DELETE_RSP 25 0 0 0 GET_STATS_RSP 0 0 0 0 STATS_TMR_EXP 0 0 0 0 TMR_EXPIRY 0 0 0 0 CREATE_STRM_REQ 0 0 0 0 MODIFY_STRM_REQ 0 0 0 0 DELETE_STRM_REQ 0 0 0 0 DETAIL_STAT_REQ 0 0 0 0 DETAIL_STAT_RSP 0 0 0 0 DT_STAT_TMR_EXP 0 0 0 0 ********************** END ACTIVE **************************** *********************** STANDBY ****************************** IDLE ALLOCATING ALLOCATED MODIFYING CREATE_REQ 0 0 0 0 MODIFY_REQ 0 0 0 0 DELETE_REQ 0 0 0 0 GET_STATS_REQ 0 0 0 0 PROV_RSP_OK 0 0 0 0 PROV_RSP_FAIL 0 0 0 0 DELETE_RSP 0 0 0 0 GET_STATS_RSP 0 0 0 0 STATS_TMR_EXP 0 0 0 0 TMR_EXPIRY 0 0 0 0 CREATE_STRM_REQ 0 0 0 0 MODIFY_STRM_REQ 0 0 0 0 DELETE_STRM_REQ 0 0 0 0 DETAIL_STAT_REQ 0 0 0 0 DETAIL_STAT_RSP 0 0 0 0 DT_STAT_TMR_EXP 0 0 0 0 DELETING ALLOC_MOD_PEND MODIFY_MOD_PEND DELETE_PENDING CREATE_REQ 0 0 0 0 MODIFY_REQ 0 0 0 0 DELETE_REQ 0 0 0 0 GET_STATS_REQ 0 0 0 0 PROV_RSP_OK 0 0 0 0 PROV_RSP_FAIL 0 0 0 0 DELETE_RSP 0 0 0 0 GET_STATS_RSP 0 0 0 0 STATS_TMR_EXP 0 0 0 0 TMR_EXPIRY 0 0 0 0 CREATE_STRM_REQ 0 0 0 0 MODIFY_STRM_REQ 0 0 0 0 DELETE_STRM_REQ 0 0 0 0 DETAIL_STAT_REQ 0 0 0 0 DETAIL_STAT_RSP 0 0 0 0 DT_STAT_TMR_EXP 0 0 0 0 ********************** END STANDBY ***************************show voip htsp
To display the voip and hybrid transport switching protocol (HTSP) connections active in the router, use the show voip htspcommand in privileged EXEC mode.
Usage Guidelines
Use theshow voip htsp command todisplay the voip and hybrid transport switching protocol (HTSP) connections active in the router.
Examples
The following is sample output from the show voip htsp command:
Router# show voip htsp NOTE: '-' means Not Applicable for that signalling type SLOT/ TSP TDM TDM PORT/ TSP BEAR CONNECT CROSS CHANNEL HTSPINFO VTSP_CDB CDB CHAN_T DONE CONNECT ========= ========== ========== ========== ========== ======= ======= 02/00/01 0x677371E8 0x68905A48 0x67757AA4 0x677371E8 y y 02/00/02 0x67737780 0x00000000 0x00000000 0x00000000 n n 02/00/03 0x67737D18 0x68906548 0x67757584 0x67737D18 y y 02/00/04 0x677382B0 0x68904C88 0x677572F4 0x677382B0 y y 02/00/05 0x67738848 0x00000000 0x00000000 0x00000000 n n 02/00/06 0x67738DE0 0x00000000 0x00000000 0x00000000 n n 02/00/07 0x67739378 0x689054C8 0x67756B44 0x67739378 y y 02/00/08 0x67739910 0x68907888 0x677568B4 0x67739910 y y 02/00/09 0x67739EA8 0x00000000 0x00000000 0x00000000 n n 02/00/10 0x6773A440 0x00000000 0x00000000 0x00000000 n n 02/00/11 0x6773A9D8 0x68906D88 0x67756104 0x6773A9D8 y y 02/00/12 0x6773AF70 0x68908388 0x67755E74 0x6773AF70 y y 02/00/13 0x6773B508 0x00000000 0x00000000 0x00000000 n n 02/00/14 0x6773BAA0 0x00000000 0x00000000 0x00000000 n n 02/00/15 0x6773C038 0x689096C8 0x677556C4 0x6773C038 y y 02/00/17 0x6773C5D0 0x68909148 0x67755434 0x6773C5D0 y y 02/00/18 0x6773CB68 0x00000000 0x00000000 0x00000000 n n 02/00/19 0x6773D100 0x00000000 0x00000000 0x00000000 n n 02/00/20 0x6773D698 0x68905788 0x67754C84 0x6773D698 y y 02/00/21 0x6773DC30 0x68905D08 0x677549F4 0x6773DC30 y y 02/00/22 0x6773E1C8 0x00000000 0x00000000 0x00000000 n n 02/00/23 0x6773E760 0x00000000 0x00000000 0x00000000 n n 02/00/24 0x6773ECF8 0x68906AC8 0x67754244 0x6773ECF8 y y 02/00/25 0x6773F290 0x68907308 0x67753FB4 0x6773F290 y y 02/00/26 0x6773F828 0x00000000 0x00000000 0x00000000 n n 02/00/27 0x6773FDC0 0x00000000 0x00000000 0x00000000 n n 02/00/28 0x67740358 0x689080C8 0x67753804 0x67740358 y y 02/00/29 0x677408F0 0x68908908 0x67753574 0x677408F0 y y 02/00/30 0x67740E88 0x00000000 0x00000000 0x00000000 n n 02/00/31 0x67741420 0x68909408 0x67753054 0x67741420 y y 02/02/01 0x67B88824 0x00000000 0x00000000 - - n 02/02/02 0x67B88DBC 0x00000000 0x00000000 - - n 02/02/03 0x67B89354 0x00000000 0x00000000 - - n 02/02/04 0x67B898EC 0x00000000 0x00000000 - - n 02/02/05 0x67B89E84 0x00000000 0x00000000 - - n 02/02/06 0x67B8A41C 0x00000000 0x00000000 - - n 02/02/07 0x67B8A9B4 0x00000000 0x00000000 - - n 02/02/08 0x67B8AF4C 0x00000000 0x00000000 - - n 02/02/09 0x67B8B4E4 0x00000000 0x00000000 - - nshow voip recmsp session
To display active recording Media Service Provider (MSP) session information, use the show voip recmsp session command in privileged EXEC mode.
Command Default
Displays brief information about recorded calls that have the anchor call ID, forked call ID, and MSP call ID.
Usage Guidelines
Use the show voip recmsp session command to display MSP-related information about the recorder, for example, the way the recording MSP views the recording session.
The show voip recmsp session detail call-id callid command provides detailed information about each recording session. It provides details about the anchor leg and nonanchor leg. It also shows how the anchor and nonanchor streams are mapped to the forked leg Real-Time Transport Protocol (RTP) streams.
Examples
The following is sample output from the show voip recmsp session detail call-id command. Fields in the display are self-explanatory.
Router# show voip recmsp session detail call-id 140 RECMSP active sessions: Detailed Information ========================= Recording MSP Leg Details: Call ID: 143 GUID : 7C5946D38ECD AnchorLeg Details: Call ID: 141 Forking Stream type: voice-nearend Participant: 708090 Non-anchor Leg Details: Call ID: 140 Forking Stream type: voice-farend Participant: 10000 Forked Leg Details: Call ID: 145 Near End Stream CallID 145 Stream State ACTIVE Far End stream CallID 146 Stream State ACTIVE Found 1 active sessionsshow voip rtp connections
To display Real-Time Transport Protocol (RTP) named event packets, use the show voip rtp connectionscommand in privileged EXEC mode.
Command History
Release
Modification
12.0
This command was introduced.
12.3(7)T
The detail keyword was added.
12.3(14)T
This command was implemented on the Cisco 2800 series and Cisco 3800 series.
12.4(2)T
This command was integrated into Cisco IOS Release 12.4(2)T.
12.4(22)T
Command output was updated to show IPv6 information.
Usage Guidelines
This command displays information about RTP named event packets, such as caller ID number, IP address, and port for both the local and remote endpoints. The output from this command provides an overview of all the connections in the system, and this information can be used to narrow the criteria for debugging. The debug voip rtp command floods the console with voice packet information. You can use the show voip rtp connections command to get caller ID, remote IP address, or remote port identifiers that you can use to limit the output from the debug voip rtp command.
The detail keyword allows you to identify the phone or phones that have connected two RTP call legs to create VoIP-to-VoIP or VoIP-to-POTS hairpins. If the detail keyword is omitted, the output does not display calls that are connected by hairpin call routing.
Examples
The table below describes the significant fields shown in the examples. Each line of output under "VoIP RTP active connections" shows information for one call leg. A phone call normally consists of two call legs, one connected to the calling party and one connected to the called party. The router joins (or bridges) the two call legs to make a call. The show voip rtp connections command shows the RTP information for H.323 and Session Initiation Protocol (SIP) calls only; it does not directly show the POTS call legs. The information for the IP phone can be seen using the show ephone offhook command.
The following sample output shows an incoming H.323 call that is being directed to an IP phone attached to a Cisco CallManager Express (CME) system.
Router# show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 21 22 16996 18174 10.4.204.37 10.4.204.24 Found 1 active RTP connectionsThe following sample output shows the same call as in the previous example, but using the detail keyword with the command. The sample output shows the called number (1509) and calling number (8108) on both call legs (21 and 22); the called and calling numbers are the same on both legs for a simple A-to-B call. Leg 21 is the H.323 segment of the and leg 22 is the POTS segment that goes to the IP phone.
Router# show voip rtp connections detail VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 21 22 16996 18174 10.4.204.37 10.4.204.24 callId 21 (dir=1):called=1509 calling=8108 redirect= dest callId 22:called=1509 calling=8108 redirect= 1 context 64FB3358 xmitFunc 6032E8B4 Found 1 active RTP connectionsThe following example shows the call from the previous example being transferred by extension 1509 to extension 1514. Notice that the dstCallId changed from 22 to 24, but the original call leg (21) for the transferred party is still present. This implies that H.450.2 capability was disabled for this particular call, because if H.450.2 was being used for the transfer, the transfer would have caused the incoming H.323 call leg to be replaced with a new call.
Router# show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 21 24 16996 18174 10.4.204.37 10.4.204.24 Found 1 active RTP connectionsThe following example shows the detailed output for the same transfer as shown in the previous example. The original incoming call leg is still present (21) and still has the original called and calling numbers. The transferred call leg (24) shows 1509 (the transferring party) as the calling party and 1514 (the transfer destination) as the called party.
Router# show voip rtp connections detail VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 21 24 16996 18174 10.4.204.37 10.4.204.24 callId 21 (dir=1):called=1509 calling=8108 redirect= dest callId 24:called=1514 calling=1509 redirect= 1 context 6466E810 xmitFunc 6032E8B4 Found 1 active RTP connectionsThe following sample output shows a cross-linked call with two H.323 call legs. The first line of output shows that the CallID for the first call leg is 7 and that this call leg is associated with another call leg that has a destination CallId of 8. The next line shows that the CallID for the leg is 8 and that it is associated with another call leg that has a destination CallId of 7. This cross-linkage between CallIds 7 and 8 shows that the first call leg is related to the second call leg (and vice versa). From this you can infer that the two call legs are actually part of the same phone call.
In an active system you can expect many lines of output that you would have to sort through to see which ones have this cross-linkage relationship. The lines showing two related call legs are not necessarily listed in adjacent order.
Router# show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 7 8 16586 22346 172.27.82.2 172.29.82.2 2 8 7 17010 16590 172.27.82.2 192.168.1.29 Found 2 active RTP connectionsThe following example shows RTP information with IPv6 local and remote addresses:
Router# show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 11 9 17424 18282 2001:DB8:C18:1:218:FEFF:FE71:2AB6 2001:DB8:C18:1:218:FEFF:FE71:2AB6 2 12 10 18282 17424 2001:DB8:C18:1:218:FEFF:FE71:2AB6 2001:DB8:C18:1:218:FEFF:FE71:2AB6 Found 2 active RTP connections
Table 6 show voip rtp connections Field Descriptions Field
Description
No.
Identifier of an RTP connection in this output.
CallId
Internal call identifier of a telephony call leg (RTP connection).
dstCallId
Internal call identifier of a VoIP call leg.
LocalRTP
RTP port of the media stream for the local entity.
RmtRTP
RTP port of the media stream for the remote entity.
LocalIP
IPv4 or IPv6 address of the media stream for the local entity.
RemoteIP
IPv4 or IPv6 address of the media stream for the remote entity.
dir
0 indicates an outgoing call. 1 indicates an incoming call.
called
Extension that received the call.
calling
Extension that made the call.
redirect
Original called number if the incoming call was forwarded.
context
Internal memory address for the control block associated with the call.
xmitFunc
Internal memory address for the transmit function to which incoming RTP packets (on the H.323 and SIP side) are sent; the address for the function that delivers the packets to the ephone.
show voip rtp forking
To display the Real-Time Transport Protocol (RTP) media-forking connections, use the show voip rtp forking command in privileged EXEC mode.
Usage Guidelines
The show voip rtp forking command displays information about RTP named event packets, such as type of stream, IP address, and port for both the local and remote endpoints. The output from this command provides an overview of all the media-forking connections in the system, and this information can be used to narrow the criteria for debugging. The debug voip rtp command floods the console with voice packet information. You can use the show voip rtp forking command to display the remote IP address, or remote port identifiers that you can use to limit the output from the debug voip rtp command.
Examples
The following is sample output from the show voip rtp forking command:
Router# show voip rtp forking VoIP RTP active forks : Fork 1 stream type voice-only (0): count 1 remote ip 9.13.36.101, remote port 20590, local port 17596 codec g711alaw, logical ssrc 0x60 packets sent 237, packets received 413 stream type voice+dtmf (1): count 0 stream type dtmf-only (2): count 0 stream type voice-nearend (3): count 1 remote ip 9.13.36.102, remote port 18226, local port 17434 codec g729r8, logical ssrc 0x103 packets sent 39, packets received 0 stream type voice+dtmf-nearend (4): count 0 stream type voice-farend (5): count 1 remote ip 9.13.36.120, remote port 16912, local port 21098 codec g729r8, logical ssrc 0x105 packets sent 39, packets received 0 stream type voice+dtmf-farend (6): count 0 stream type video (7): count 0The table below describes the significant fields shown in the display.
Table 7 show voip rtp forking Field Descriptions Field
Description
stream type
Indicates the type of stream.
count
Number of packets in the specified type of stream.
remote ip
IPv4 or IPv6 address of the media stream for the remote entity.
remote port
RTP port of the media stream for the remote entity.
local port
RTP port of the media stream for the local entity.
codec
Codec supported on the specified channel.
logical ssrc
Indicates the logical synchronization source (SSRC) for the specified channel.
packets sent
Total number of packets sent from the channel.
packets received
Total number of packets received by the channel.
show voip trunk group
To display the internal list of voip trunk groups, use the show voip trunk group command in user EXEC or privileged EXEC mode.
show vrm active_calls
To display active-only voice calls either for a specific voice feature card (VFC) or for all VFCs, use the show vrm active_callscommand in privileged EXEC mode.
Usage Guidelines
Use this command to display active-only voice calls either for a specific VFC or for all VFCs. Each active call occupies a block of information describing the call. This information provides basically the same information as the show vrm vdevice command.
Examples
The following is sample output from this command specifying a dial-shelf slot number:
Router# show vrm active_calls 6 slot = 6 virtual voice dev (tag) = 61 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 241 Resource (vdev_common) status = 401 means :active others tot ingress data = 24 tot ingress control = 1308 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 22051 tot egress control = 1304 tot egress data drops = 0 tot egress control drops = 0 slot = 6 virtual voice dev (tag) = 40 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 157 Resource (vdev_common) status = 401 means :active othersThe table below describes significant fields shown in this output.
show vrm vdevices
To display detailed information for a specific digital signal processor (DSP) or summary information for all voice feature cards (VFCs), use the show vrm vdevicescommand in privileged EXEC mode.
show vrm vdevices { vfc-slot-number voice-device-number | alarms [vfc-slot-number-for-alarms] | summary }
Syntax Description
vfc -slot-number
Slot number of the VFC. Range is from 0 to 11.
voice -device-number
DSP number. Range is from 1 to 96.
alarms
DSP alarm statistics for all DSPs on all slots or specified slots.
vfc -slot-number-for-alarms
(Optional) Slots for which you need alarm information. If no slots are specified, alarm information for all slots is displayed.
summary
Synopsis of voice feature card DSP mappings, capabilities, and resource states.
Usage Guidelines
Use this command to display detailed information for a specific DSP or a brief summary for all VFCs. The display provides information such as the number of channels, channels per DSP, bitmap of digital signal processor modules (DSPMs), DSP alarm statistics, and version numbers. This information is useful in monitoring the current state of your VFCs.
The display for a specific DSP provides information on the codec that each channel is using, if active, or on the codec that was last used and whether the channel is not currently sending cells. It also displays the state of the resource. In most cases, if there is an active call on that channel, the resource should be marked active. If the resource is marked as reset or bad, this may be an indication of a response loss for the VFC on a reset request. If this condition persists, you might experience a problem with the communication link between the router shelf and the VFC.
Examples
The following is sample output from this command specifying dial-shelf slot number and DSP number. In this particular example, the call is active so the statistics displayed are for this active call. If no calls are currently active on the device, the statistics would be for the previous (or last active) call.
Router# show vrm vdevices 6 1 slot = 6 virtual voice dev (tag) = 1 channel id = 1 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 0 Resource (vdev_common) status = 401 means :active others tot ingress data = 101 tot ingress control = 1194 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 39722 tot egress control = 1209 tot egress data drops = 0 tot egress control drops = 0 slot = 6 virtual voice dev (tag) = 1 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 1 Resource (vdev_common) status = 401 means :active others tot ingress data = 21 tot ingress control = 1167 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 19476 tot egress control = 1163 tot egress data drops = 0 tot egress control drops = 0The table below describes significant fields shown in this output.
The following sample output displays alarm statistics for slot 6 of the DSP.
Router# show vrm vdevices alarms 6 ----------------------ALARM STATISTICS FOR SLOT 6 ------------------------ TAG Mod DSP Chn OperStat AlmCnt AlmTime AlmCause AlmText -------------------------------------------------------------------------- 1 1 1 1 READY CD 0 0 1 2 READY CD 0 0 1 2 1 2 1 READY CD 0 0 1 2 READY CD 0 0 1 3 1 3 1 READY CD 0 0 1 2 READY CD 0 0 1 4 1 4 1 READY CD 0 0 1 2 READY CD 0 0 1 5 1 5 1 READY CD 0 0 1 2 READY CD 0 0 1 6 1 6 1 READY CD 0 0 1 2 READY CD 0 0 1 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 7 2 1 1 READY CD 0 0 1 2 READY CD 0 0 1 8 2 2 1 READY CD 0 0 1 2 READY CD 0 0 1 9 2 3 1 READY CD 0 0 1 2 READY CD 0 0 1 10 2 4 1 READY CD 0 0 1 ! 94 16 4 1 READY CD 0 0 1 2 READY CD 0 0 1 95 16 5 1 READY CD 0 0 1 2 READY CD 0 0 1 96 16 6 1 READY CD 0 0 1 2 READY CD 0 0 1 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++The table below describes significant fields shown in this output.
Table 10 show vrm vdevices alarms Field Descriptions Field
Description
TAG
Logical tag number.
Mod
DSP module number.
DSP
DSP number within the module.
Chn
Channel number for the DSP within the module.
OperStat
Operational status of the channel.
AlmCnt
Alarm count since bootup on that channel.
AlmTime
Time at which last alarm message was received.
AlmCause
Cause of last alarm message received.
AlmText
Text message corresponding to the last alarm message.
Possible Values for the Operational Status of the Channel (OperStat)
RESET
RESET state.
DOWN
DOWN state.
READY CR
CORE READY state.
READY CD
CODEC READY state.
IDLE V
VOICE IDLE state.
IDLE FAX
FAX IDLE state.
READY V
VOICE READY state.
READY FX
FAX READY state.
READY D
DTMF READY state.
UNKNOWN
UNKNOWN state.
The following is sample output from this command specifying a summary list. In the "Voice Device Mapping" area, the "C_Ac" column indicates the number of active calls for a specific DSP. If there are any nonzero numbers under the "C_Rst" and/or "C_Bad" column, a reset request was sent, but it was lost; this could mean a faulty DSP.
Router# show vrm vdevices summary *********************************************************** ******summary of voice devices for all voice cards********* *********************************************************** slot = 6 major ver = 0 minor ver = 1 core type used = 2 number of modules = 16 number of voice devices (DSPs) = 96 chans per vdevice = 2 tot chans = 192 tot active calls = 178 module presense bit map = FFFF tdm mode = 1 num_of_tdm_timeslots = 384 auto recovery is on number of default voice file (core type images) = 2 file 0 maj ver = 0 min ver = 0 core_type = 1 trough size = 2880 slop value = 0 built-in codec bitmap = 0 loadable codec bitmap = 0 fax codec bitmap = 0 file 1 maj ver = 3 min ver = 1 core_type = 2 trough size = 2880 slop value = 1440 built-in codec bitmap = 40B loadable codec bitmap = BFC fax codec bitmap = 7E -------------------Voice Device Mapping------------------------ Logical Device (Tag) Module# DSP# C_Ac C_Busy C_Rst C_Bad --------------------------------------------------------------- 1 1 1 2 0 0 0 2 1 2 2 0 0 0 3 1 3 2 0 0 0 4 1 4 2 0 0 0 5 1 5 2 0 0 0 6 1 6 2 0 0 0 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 7 2 1 2 0 0 0 8 2 2 2 0 0 0 9 2 3 2 0 0 0 10 2 4 1 0 0 0 11 2 5 2 0 0 0 12 2 6 1 0 0 0 . . . 91 16 1 2 0 0 0 92 16 2 2 0 0 0 93 16 3 1 0 0 0 94 16 4 2 0 0 0 95 16 5 2 0 0 0 96 16 6 2 0 0 0 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Total active call channels = 178 Total busied out channels = 0 Total channels in reset = 0 Total bad channels = 0 Note :Channels could be in multiple statesThe table below describes significant fields shown in this output.
show vsp
To display cumulative information about voice streaming processing (VSP) sessions, use the show vsp command in privileged EXEC mode.
Syntax Description
all
Displays all available information on VSP sessions, including the information specified by the other keywords listed in this table.
debug
Displays the type of debugging information that is enabled by using the debug vsp command.
session
Displays cumulative statistics about active VSP sessions.
statistics
Displays statistics about active VSP sessions, including memory statistics.
Usage Guidelines
Use the clear vsp statistics command to reset the counters to 0 for the show vsp command.
Examples
The following is sample output from the show vsp debug command:
Router# show vsp debug VSP:<1>[0x62291660](0x62291660) debug_flag=0x7FFThe following is sample output from the show vsp session command:
Router# show vsp session VSP_STATS:Session Statistics - sessions total=0; max_active=0, current=0 session_duration last=0; max=0, min=0 ms pre_stream_wait last=0; max=0, min=0 ms stream_duration last=0; max=0, min=0 ms post_stream_wait last=0; max=0, min=0 ms stream_size last=0; max=0, min=0 bytes streaming_rate last=0; max=0, min=0 bytes/sec total_packet_count last=0; max=0, min=0 packets drop_packet_count last=0; max=0, min=0 packets particle_packet_count last=0; max=0, min=0 packetsThe following is sample output from the show vsp statistics command:
Router# show vsp statistics VSP_STATS:Session Statistics - sessions total=0; max_active=0, current=0 session_duration last=0; max=0, min=0 ms pre_stream_wait last=0; max=0, min=0 ms stream_duration last=0; max=0, min=0 ms post_stream_wait last=0; max=0, min=0 ms stream_size last=0; max=0, min=0 bytes streaming_rate last=0; max=0, min=0 bytes/sec total_packet_count last=0; max=0, min=0 packets drop_packet_count last=0; max=0, min=0 packets particle_packet_count last=0; max=0, min=0 packets VSP_STATS: Format Statistics - au_format_count=20 wav_format_count=3 other_format_count=0 VSP_STATS: Codec Statistics - codec_g729_count=4 codec_g726_count=10 codec_g711_count=0 codec_g728_count=2 codec_g723_count=5 codec_gsm_count=2 codec_other_count=0 VSP_STATS: Media Statistics - ram_count=23 http_count=0 smtp_count=0 rtsp_count=0 other_count=0 VSP_STATS:RTP Statistics - ts_gap_samples max=76800, min=80 samples [Unexpected SSRC Change (USC)] usc_count last=0; total=0, max=0, min=0 [Out of sequence packet (OOSP)] oosp_count last=0; total=0, max=0, min=0 [Unexpected timestamp gap (UTG)] max_utg_count last=0; total=0, max=0, min=0 [Comfort Noise (CN)] max_cn_count last=4; total=70, max=8, min=4 [Unexpected payload type or size (UPTS)] upt_count last=0; total=0, max=0, min=0; last_type=0 ups_count last=0; total=198, max=61, min=0; last_size=2 bytes [Data exceeds limit (DEL)] del_count last=0; total=2, max=1, min=0 [Silence exceeds timeout (SET)] set_count last=0; total=0, max=0, min=0 VSP_STATS:Packet Statistics - [Silence patching total (SPT)] spt_count last=296; total=7230, max=889, min=290 [Concealment patching total (CPT)] cpt_count last=0; total=34, max=18, min=0 [Normal patching total (NPT)] npt_count last=171; total=4249, max=453, min=106The table below describes the fields shown in this output.
Table 12 show vsp statistics Field Descriptions Field
Description
Session Statistics
sessions total; max_active, current
Total number of VSP sessions since router startup or since the clear vsp statistics command was used. The active value should always be 0.
session_duration last; max, min
Duration of the last (most recent) session, and of the longest and shortest sessions in msecs.
pre_stream_wait last; max, min
Msecs that elapsed before the arrival of the first packet. Values are shown for last session, and for the session with the longest and shortest waits.
stream_duration last; max, min
Msecs between first packet arrival and last packet flush. Values are shown for last session, and for the session with the longest and shortest durations.
post_stream_wait last; max, min
Msecs between last packet flush and close of session.
stream_size last; max, min
Data streaming size.
streaming_rate last; max, min
Data streaming rate.
total_packet_count last; max, min
Total packets processed.
drop_packet_count last; max, min
Total packets dropped. The difference between the total packet count and packets dropped is the number of packets that have been accepted.
particle_packet_count last; max, min
Total particle packets processed.
Format Statistics
au_format_count
Number of VSP sessions that used audio files in .au format.
wav_format_count
Number of VSP sessions that used audio files in .wav format.
other_format_count
Number of VSP sessions that used audio files of an unknown format.
Codec Statistics
codec_g729_count
Number of VSP sessions that used the G.729 codec.
codec_g726_count
Number of VSP sessions that used the G.726 codec.
codec_g711_count
Number of VSP sessions that used the G.711 codec.
codec_g728_count
Number of VSP sessions that used the G.728 codec.
codec_g723_count
Number of VSP sessions that used the G.723 codec.
codec_gsm_count
Number of VSP sessions that used the GSM codec.
codec_other_count
Number of VSP sessions that used an unknown codec.
Media Statistics
ram_count
Total number of RAM recordings and playouts.
http_count
Total number of HTTP recordings and playouts.
smtp_count
Total number of SMTP recordings.
rtsp_count
Total number of RTSP recordings and playouts.
other_count
Should always be 0.
RTP Statistics
ts_gap_samples max min
Permissible timestamp gap in samples.
[Unexpected SSRC Change (USC)]
usc_count last; total, max, min
Number of times that the source of the streaming has changed.
[Out of sequence packet (OOSP)]
oosp_count last; total, max, min
Number of out-of-sequence packets.
[Unexpected timestamp gap (UTG)]
max_utg_count last; total, max, min
Number of packets with an unexpected timestamp gap.
[Unexpected payload type or size (UPTS)]
upt_count last; total, max, min; last_type
Number of comfort noise packets.
ups_count last; total, max, min; last_size
Number of packets with unexpected nonvoice payload sizes.
[Data exceeds limit (DEL)]
del_count last; total, max, min
Number of times that the total recording size is larger than the preset recording size.
[Silence exceeds timeout (SET)]
set_count last; total, max, min
Number of times that the timestamp gap is larger than the preset timeout value.
Packet Statistics
[Silence patching total (SPT)]
spt_count last; total, max, min
Number of silence packets that have been inserted during recording.
[Concealment patching total (CPT)]
cpt_count last; total, max, min
Number of concealment packets that have been inserted during recording.
[Normal patching total (NPT)]
npt_count last; total, max, min
Number of normal packets that have been patched during recording.
show wsapi
To display information on the Cisco Unified Communication IOS services, including registration, statistics, and route information, use the show wsapi command in user EXEC or privileged EXEC mode.
show wsapi { http-client | http-server | registration | registration { all | xcc | xcdr | xsvc } | svcc route }
Syntax Description
http-client
Displays the statistics that have been collected on the http client interface.
http-server
Displays the statistics that have been collected on the http server interface.
registration
Displays the currently registered applications on the WSAPI subsystem.
all
Displays all registered applications.
xcc
Displays the applications that are registered to the XCC provider.
xcdr
Displays the applications that are registered to the XCDR provider.
xsvc
Displays the applications that are registered to the XSVC provider.
xsvc route
Displays the internal route information in the XSVC provider.
Usage Guidelines
Use this command to display information on the Cisco Unified Communication IOS services.
Examples
The following example shows a sample output from the show wsapi http-client command.
Router# show wsapi http-client WSAPI Outgoing Notify/Solicit Message Statistics ========================================= wsapi_show_httpc_callback_context_invalid: 0 wsapi_show_httpc_callback_context_error: 0 wsapi_show_httpc_callback_no_reg: 5 wsapi_show_httpc_callback_notify_OK: 85 wsapi_show_httpc_callback_notify_error: 0 wsapi_show_httpc_callback_client_error: 0 wsapi_show_httpc_callback_error: 7 wsapi_show_httpc_callback_client_error: 0 wsapi_show_httpc_callback_decode_error: 28 wsapi_show_httpc_callback_no_txID: 0 wsapi_show_httpc_callback_OK: 655 wsapi_show_httpc_create_msg_error: 0 wsapi_show_httpc_context_active: 0 wsapi_tx_context_freeq depth: 4The following example shows a sample output from the show wsapi http-server command.
Router# show wsapi http-server WSAPI Incoming Request Message Statistics ========================================= wsapi_show_https_urlhook: 23 wsapi_show_https_post_action: 23 wsapi_show_https_post_action_fail: 0 wsapi_show_https_xml_fault: 0 wsapi_show_https_post_action_done: 23 wsapi_show_https_service_timeout: 0 wsapi_show_https_send_error: 0 wsapi_show_https_invalid_context: 0 wsapi_show_https_data_active: 0 wsapi_https_data_q depth: 1 wsapi_show_https_internal_service_error: 0 wsapi_show_https_service_unavailable_503: 0 wsapi_show_https_not_found_404: 0 wsapi_show_https_registration_success: 9 wsapi_show_https_not_registered: 0 wsapi_show_https_registration_auth_fail: 1 wsapi_show_https_registration_fail: 0 wsapi_show_https_un_registered: 0The following example shows a sample output from the show wsapi registration command.
Router# show wsapi registration Provider XCC ===================================================== registration id: 4FA11CC:XCC:myapp:5 appUrl:http://sj22lab-as2:8090/xcc appName: myapp provUrl: http://10.1.1.1:8090/cisco_xcc prober state: STEADY connEventsFilter: CREATED|AUTHORIZE_CALL|ADDRESS_ANALYZE|REDIRECTED|ALERTING|CONNECTED|TRANSFERRED|CALL_DELIVERY|DISCONNECTED|HANDOFF_JOIN|HANDOFF_LEAVE mediaEventsFilter: DTMF|MEDIA_ACTIVITY|MODE_CHANGE||TONE_DIAL|TONE_OUT_OF_SERVICE|TONE_RINGBACK|TONE_SECOND_DIAL blockingEventTimeoutSec: 1 blockingTimeoutHandle: CONTINUE_PROCESSING Provider XSVC ===================================================== registration index: 2 id: 4FA0F8C:XSVC:myapp:3 appUrl:http://sj22lab-as2:8090/xsvc appName: myapp provUrl: http://10.1.1.1:8090/cisco_xsvc prober state: STEADY route filter: event filter: off Provider XCDR ===================================================== registration index: 1 id: 4FA10A0:XCDR:myapp:1 appUrl:http://sj22lab-as2:8090/xcdr appName: myapp provUrl: http://10.1.1.1:8090/cisco_xcdr prober state: STEADY cdr format: COMPACT event filter: offThe following example shows a sample output from the show wsapi xsvc route command.
Router# show wsapi xsvc route Route SANJOSE_SIP ===================================================== Type: VOIP Description: OUT Filter: Trunk: Trunk Name: 1.3.45.2 Trunk Type: SIPV2 Trunk Status: UP Route SANJOSE_PRI ===================================================== Type: PSTN Description: IN Filter: Trunk: Trunk Name: Se0/1/0:23 Trunk Type: ISDN PRI Trunk Status: UP Total channels 2 Channel bitmap 0x01FFFFFE 1-24 Link bitmap 0x00000006 Alarm 0x00000001 Time elapsed 516 Interval 92 CurrentData 0 Line Code Violations, 0 Path Code Violations 0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs TotalData 49 Line Code Violations, 7 Path Code Violations, 0 Slip Secs, 1 Fr Loss Secs, 1 Line Err Secs, 0 Degraded Mins, 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 2 Unavail Secs Trunk Name: Se0/1/1:23 Trunk Type: ISDN PRI Trunk Status: UP Total channels 2 Channel bitmap 0x01FFFFFE 1-24 Link bitmap 0x00000006 Alarm 0x00000001 Time elapsed 516 Interval 92 CurrentData 0 Line Code Violations, 0 Path Code Violations 0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs TotalData 42 Line Code Violations, 4 Path Code Violations, 0 Slip Secs, 1 Fr Loss Secs, 1 Line Err Secs, 0 Degraded Mins, 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 2 Unavail Secsshow xcsp port
To display the status of a router port under the control of the external control service provider (XCSP) subsystem, use the show xcsp portcommand in privileged EXEC mode.
Syntax Description
Examples
The following is sample output from this command:
Router# show xcsp port 1 0 Slot 1 configured Number of ports configured=1 slot state= Up =================================================== Port 0 State= Up type = 5850 24 port T1 Channel states 0 Idle 1 Idle 2 Idle 3 Idle 4 Idle . . . 22 Idle 23 IdleThe table below describes significant fields in this output.
NoteTo get the field description output, you must enter the slot-num and port-num arguments for the show xcsp port command.
Table 13 show xcsp port Field Descriptions Field
Descriptions
Port
Port number. Range is from 1 to 28.
State
Port state; can be Up or Down.
type
T1 or E1 ports on the AS5400: 8. T1 or E1 ports on the AS5850: 24. T3 ports on the AS5400 and AS5850: 28.
Channel states
Channel states. Values are as follows:
show xcsp slot
To display the status of a router slot under the control of the external control service provider (XCSP) subsystem, use the show xcsp slot command in privileged EXEC mode.
shut
shutdown (Annex G neighbor)
To disable the service relationships requirement for border elements, use the shutdown command in config-nxg-neigh-srvc mode. To enable the service relationship for border elements, use the noform of this command.
Usage Guidelines
The no shutdown command verifies that a domain name has been configured and ensures that the border element has been configured to reject messages from unknown "stranger" border elements.
Examples
The following example enables the border element:
Router(config-nxg-neigh-srvc)# no shutdown
Related Commands
Command
Description
access -policy
Requires that a neighbor be explicitly configured.
inbound ttl
Sets the inbound time-to-live value.
outbound retry -interval
Defines the retry period for attempting to establish the outbound relationship between border elements.
retry interval
Defines the time between delivery attempts.
retry window
Defines the total time that a border element attempts delivery.
shutdown (Annex G)
To shut down the Annex G border element (BE), use the shutdown command in Annex G configuration mode. To reinstate the Annex G BE, use the no form of this command.
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. This command was not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
Usage Guidelines
While the Annex G BE is in shutdown state, all Annex G messages received from neighbors are ignored and the colocated gatekeeper does not use the Annex G BE for address resolution.
shutdown (dial-peer)
To change the administrative state of the selected dial peer from up to down, use the shutdown command in dial-peer configuration mode. To change the administrative state of this dial peer from down to up, use the no form of this command.
Usage Guidelines
When a dial peer is shut down, you cannot initiate calls to that peer.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example changes the administrative state of voice telephony (plain old telephone service [POTS]) dial peer 10 to down:
dial-peer voice 10 pots shutdownThe following example changes the administrative state of voice telephony (POTS) dial peer 10 to up:
dial-peer voice 10 pots no shutdownshutdown (DSP Farm profile)
To disable the digital signal processor (DSP) farm profile, use the shutdown command in DSP farm profile configuration mode. To allocate DSP farm resources and associate with the application, use the no form of this command.
Usage Guidelines
It is essential that the profile be disabled by using the shutdown command before a DSP farm profile is updated.
Examples
The following example allocates DSP farm resources and associates with the application:
Router(config-dspfarm-profile)# no shutdownRelated Commands
Command
Description
codec (dspfarm-profile)
Specifies the codecs supported by a DSP farm profile.
description (dspfarm-profile)
Includes a specific description about the DSP farm profile.
dspfarm profile
Enters the DSP farm profile configuration mode and defines a profile for DSP farm services.
maximum sessions (dspfarm-profile)
Specifies the maximum number of sessions that need to be supported by the profile.
shutdown (gatekeeper)
To disable the gatekeeper, use the shutdown command in gatekeeper configuration mode. To enable the gatekeeper, use the no form of this command.
Usage Guidelines
The gatekeeper does not have to be enabled before you can use the other gatekeeper configuration commands. In fact, it is recommended that you complete the gatekeeper configuration before bringing up the gatekeeper because some characteristics may be difficult to alter while the gatekeeper is running, as there may be active registrations or calls.
The no shutdown command enables the gatekeeper, but it does not make the gatekeeper operational. The two exceptions to this are as follows:
- If no local zones are configured, a no shutdown command places the gatekeeper in INACTIVE mode waiting for a local zone definition.
- If local zones are defined to use an HSRP virtual address, and the HSRP interface is in STANDBY mode, the gatekeeper goes into HSRP STANDBY mode. Only when the HSRP interface is ACTIVE does the gatekeeper go into the operational UP mode.
shutdown (gateway)
To shut down all VoIP call service on a gateway, use the shutdowncommandin voice service configuration mode. To enable VoIP call service, use the no form of this command.
shutdown (mediacard)
To disable a selected media card, use the shutdown command in mediacard configuration mode. To enable a selected media card, use the no form of this command.
Usage Guidelines
Use the no shutdown command at the end of media card configuration. If there are any active connections when you disable the media card, the Digital Signal Processor Resource Manager (DSPRM) displays a warning message indicating that the DSP resources allocated on other media cards for some of the resource pool in this media card will be removed or that there are active connections available in this resource pool and prompts you for a response. Profiles that use resources on this card must be brought up separately after using this command.
shutdown (auto-config application)
To disable an auto-configuration application for download, use the shutdown command in auto-config application configuration mode. To enable an auto-configuration application for download, use the noform of this command.
shutdown (RLM)
To shut down all of the links under the RLM group, use the shutdown command in RLM configuration mode. RLM does not try to reestablish those links until the command is negated. To disable this function, use the no form of this command.
Related Commands
Command
Description
clear interface
Resets the hardware logic on an interface.
clear rlm group
Clears all RLM group time stamps to zero.
interface
Defines the IP addresses of the server, configures an interface type, and enters interface configuration mode.
link (RLM)
Specifies the link preference.
protocol rlm port
Reconfigures the port number for the basic RLM connection for the whole rlm-group.
retry keepalive
Allows consecutive keepalive failures a certain amount of time before the link is declared down.
server (RLM)
Defines the IP addresses of the server.
show rlm group statistics
Displays the network latency of the RLM group.
show rlm group status
Displays the status of the RLM group.
show rlm group timer
Displays the current RLM group timer values.
timer
Overwrites the default setting of timeout values.
shutdown (settlement)
To deactivate the settlement provider, use the shutdown command in settlement configuration mode. To activate a settlement provider, use the no shutdown command.
Command Default
The default status of a settlement provider is deactivated. The settlement provider is down.
Usage Guidelines
Use this command at the end of the configuration of a settlement server to bring up the provider. This command activates the provider. Otherwise, transactions do not go through the provider to be audited and charged. Use the shutdown command to deactivate the provider.
Examples
The following example enables a settlement server:
settlement 0 no shutdownThe following example disables a settlement server:
settlement 0 shutdownRelated Commands
Command
Description
connection -timeout
Configures the time that a connection is maintained after completing a communication exchange.
customer -id
Identifies a carrier or ISP with a settlement provider.
device -id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max -connection
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
response -timeout
Configures the maximum time to wait for a response from a server.
retry -delay
Sets the time between attempts to connect with the settlement provider.
session -timeout
Sets the interval for closing the connection when there is no input or output traffic.
settlement
Enters settlement configuration mode and specifies the attributes specific to a settlement provider.
type
Configures an SAA-RTR operation type.
shutdown (voice-port)
To take the voice ports for a specific voice interface card offline, use the shutdown command in voice-port configuration mode. To put the ports back in service, use the no form of this command.
