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This command has no arguments or keywords.
The default for this command is enabled for all voice-port types.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
This command is associated with the echo canceller operation. The echo-cancel enable command must be enabled for the non-linear command to take effect. Use the non-linear command to shut off any signal if no near-end speech is detected.
Enabling the non-linear command normally improves performance, although some users might perceive truncation of consonants at the end of sentences when this command is enabled.
This feature is also known as residual echo suppression.
The following example enables nonlinear call processing:
voice-port 0:D non-linear
echo-cancel enable
extension-number | Intger(s) defining an extension number for a particular dial peer. |
expanded-number | Integer(s) defining the expanded telephone number or destination pattern for the extension number listed. |
Global configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the num-exp global configuration command to define how to expand a particular set of numbers (for example, an extension number) into a particular destination pattern. With this command, you can map specific extensions and expanded numbers together by explicitly defining each number, or you can define extensions and expanded numbers using variables. You can also use this command to convert seven-digit numbers to numbers containing fewer than seven digits.
Use a period (.) as a variable or wildcard, representing a single number. Use a separate period for each number you want to represent with a wildcard---meaning that if you want to replace four numbers in an extension with wildcards, use four periods.
The following example expands the extension number 65541 to be expanded to 14085665541:
num-exp 65541 14085665541
The following example shows how to expand all five-digit extensions beginning with 6 to append the following numbers at the beginning of the extension number 1408566:
num-exp 6.... 1408566....
value | Specifies, in decibels, the amount of attenuation at the transmit side of the interface. Acceptable values are any integer from 0 to 14. |
The default value for T1 and E1 ports is 0.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
A system-wide loss plan must be implemented using both input gain and output attenuation commands. Other equipment (including PBXs) in the system must be taken into account when creating a loss plan. The default value for this command assumes that a standard transmission loss plan is in effect, meaning that normally, there must be -6 dB attenuation between the phones. Connections are implemented to provide -6 dB of attenuation when the input gain and output attenuation commands are configured with the default value of 0.
Please note that you cannot increase the gain of a signal going out into the PSTN, but you can decrease it. Therefore, if the voice level is too high, you can decrease the volume by either decreasing the input gain value or by increasing the output attenuation.
You can increase the gain of a signal coming in to the router. If the voice level is too low, you can increase the input gain by using the input gain command.
The following example configures a 3-decibel gain to be inserted at the transmit side of the interface:
voice-port 0:D output attenuation 3
input gain
controller number:D | Specifies the T1 or E1 controller; :D indicates the D channel associated with ISDN PRI. |
No port is configured.
Dial-peer configuration
This command first appeared in Cisco IOS Release 11.3 T.
This command is used for calls incoming from a telephony interface to select an incoming dial peer and for calls coming from the VoIP network to match a port with the selected outgoing dial peer.
This command applies only to POTS peers.
The following example associates POTS dial peer 10 with voice port 0:D:
dial-peer voice 10 pots port 0:D
string | Integers representing the prefix of the telephone number associated with the specified dial peer. Valid numbers are 0 through 9, and a comma (,). Use a comma to include a pause in the prefix. |
The default for this command is a null string.
Dial-peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the prefix command to specify a prefix for a specific dial peer. When an outgoing call is initiated to this dial peer, the prefix string value is sent to the telephony interface first, before the telephone number associated with the dial peer.
If you want to configure different prefixes for dialed numbers on the same interface, you need to configure different dial peers.
This command applies only to POTS peers.
The following example specifies a prefix of "9" and then a pause:
dial-peer voice 10 pots prefix 9,
answer-address
destination-pattern
best-effort | Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation. |
controlled-load | Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to ensure that preferential service is received even when the bandwidth is overloaded. |
guaranteed-delay | Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded. |
The default value for this command is best-effort. The no form of this command restores the default value.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the req-qos command to request a specific quality of service to be used in reaching a dial peer. Like acc-qos, when you issue this command, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses RSVP to request quality of service guarantees from the network.
This command applies only to VoIP peers.
The following example configures guaranteed-delay as the desired quality of service to a dial peer:
dial-peer voice 10 voip req-qos guaranteed-delay
acc-qos
cisco | Specifies Cisco session protocol. |
The default value for this command is cisco.
Dial-peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
For this release, Cisco session protocol (cisco) is the only applicable session protocol.
This command applies only to VoIP peers.
The following example selects Cisco session protocol as the session protocol:
dial-peer voice 10 voip session protocol cisco
session target
ipv4:destination-address | IP address of the dial peer. |
dns:host-name | Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. (Optional) You can use one of the following three wildcards with this keyword when defining the session target for VoIP peers: · $s$.---Indicates that the source destination pattern will be used as part of the domain name. · $d$.---Indicates that the destination number will be used as part of the domain name. · $u$.---Indicates that the unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name. · $e$.---Indicates that the destination pattern is used as part of the domain name in reverse dotted format for tpc.int DNS format. For example, if the destination number is 310 555-1234 and the session target is configured as $e$.cisco.com, the translated DNS name will be 4.3.2.1.5.5.5.0.1.3.cisco.com. |
loopback:rtp | Indicates that all voice data will be looped back to the originating source. This applies toVoIP peers. |
loopback:compressed | Indicates that all voice data will be looped back in compressed mode to the originating source. This applies to POTS peers. |
loopback:uncompressed | Indicates that all voice data will be looped back in uncompressed mode to the originating source. This applies to POTS peers. |
The default for this command is enabled with no IP address or domain name defined.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origination and the loopback type selected.
The session target dns command can be used with or without the specified wildcards. Using the optional wildcards can reduce the number of VoIP dial peer session targets you need to configure if you have groups of numbers associated with a particular router.
The following example configures a session target using dns for a host, "voice_router," in the domain "cisco.com":
dial-peer voice 10 voip session target dns:voice_router.cisco.com
The following example configures a session target using DNS, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number---in this case, the four-digit extension, to identify the dial peer. As in the previous example, the domain is "cisco.com."
dial-peer voice 10 voip destination-pattern 1310222.... session target dns:$u$.cisco.com
The following example configures a session target using DNS, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13105551111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voip destination-pattern 13105551111 session target dns:$d$.cisco.com
destination-pattern
session protocol
To show the active call table, use the show call active voice privileged EXEC command.
show call active voiceThis command contains no arguments or keywords.
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show call active voice privileged EXEC command to display the contents of the active call table, which shows all of the calls currently connected through the router.
For each call, there are two call legs, usually a POTS call leg and a VoIP call leg. A call leg is a discrete segment of a call connection that lies between two points in the connection. Each dial peer creates a call leg, as shown in Figure 12.
These two call legs are associated by the connection ID. The connection ID is global across the voice network, so that you can associate two call legs on one router with two call legs on another router, thereby providing an end-to-end view of a call.
The following is sample output from the show call active voice command:
router#show call active voice GENERIC: SetupTime=179388054 ms Index=1 PeerAddress=+5.... PeerSubAddress= PeerId=5 PeerIfIndex=32 LogicalIfIndex=29 ConnectTime=179389793 ms CallState=4 CallOrigin=2 ChargedUnits=0 InfoType=2 TransmitPackets=532 TransmitBytes=10640 ReceivePackets=147 ReceiveBytes=2940 TELE: ConnectionId=[0xE3EA3FF8 0xFF6D0105 0x0 0x6AEC71E4] TxDuration=23230 ms VoiceTxDuration=2940 ms FaxTxDuration=0 ms CoderTypeRate=g729r8 NoiseLevel=-84 ACOMLevel=20 OutSignalLevel=-66 InSignalLevel=-66 InfoActivity=2 ERLLevel=20 SessionTarget= GENERIC: SetupTime=179388237 ms Index=1 PeerAddress=+3622 PeerSubAddress= PeerId=3 PeerIfIndex=31 LogicalIfIndex=0 ConnectTime=179389793 ms CallState=4 CallOrigin=1 ChargedUnits=0 InfoType=2 TransmitPackets=143 TransmitBytes=2860 ReceivePackets=580 ReceiveBytes=11600 VOIP: ConnectionId[0xE3EA3FF8 0xFF6D0105 0x0 0x6AEC71E4] RemoteIPAddress=172.24.96.200 RemoteUDPPort=16422 RoundTripDelay=37 ms SelectedQoS=best-effort SessionProtocol=cisco SessionTarget=ipv4:172.24.96.200 OnTimeRvPlayout=9920 GapFillWithSilence=0 ms GapFillWithPrediction=0 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=70 ms LoWaterPlayoutDelay=30 ms ReceiveDelay=30 ms VAD = enabled CoderTypeRate=g729r8
Table 7 provides an alphabetical listing of the possible show call active voice fields and a description of each field.
| Field | Description |
|---|---|
ACOM Level | Current ACOM level for this call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for this call. |
CallOrigin | Call origin: answer or originate. |
CallState | Current state of the call. |
CoderTypeRate | Negotiated coder transmit rate of voice/fax compression during this call. |
ConnectionId | Global call identifier for this gateway call. |
ConnectTime | Time at which the call was connected. |
Dial-Peer | Tag of the dial peer transmitting this call. |
ERLLevel | Current Echo Return Loss (ERL) level for this call. |
FaxTxDuration | Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value. |
GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received in time from voice gateway for this call. |
GapFillWith Redundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because voice data was lost or not received in time from voice gateway for this call. |
GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. Examples of such pullout are frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms. |
GapFillWith Silence | Duration of voice signal replaced with silence because voice data was lost or not received in time for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during this call. |
Index | Dial-peer identification number. |
InfoActivity | Active information transfer activity state for this call. |
InfoType | Information type for this call. |
InSignalLevel | Active input signal level from the telephony interface used by this call. |
LogicalIfIndex | Index number of the logical interface for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during this call. |
NoiseLevel | Active noise level for this call. |
OnTimeRvPlayout | Duration of voice playout from data received in time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
OutSignalLevel | Active output signal level to telephony interface used by this call. |
PeerAddress | Destination pattern associated with this peer. |
PeerId | ID value of the peer table entry to which this call was made. |
PeerIfIndex | Voice-port index number for this peer. |
PeerSubaddress | Subaddress to which this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the Decoder Delay during this call. |
ReceivePackets | Number of packets received by this peer during this call. |
RemoteIPAddress | Remote system IP address for the VoIP call. |
RemoteUDPPort | Remote system UDP listener port to which voice packets are transmitted. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone during this call. |
SelectedQoS | Selected RSVP quality of service (QoS) for this call. |
SessionProtocol | Session protocol used for an Internet call between the local and remote router via the IP backbone. |
SessionTarget | Session target of the peer used for this call. |
SetupTime | Value of the system UpTime when the call associated with this entry was started. |
TransmitBytes | Number of bytes transmitted from this peer during this call. |
TransmitPackets | Number of packets transmitted from this peer during this call. |
TxDuration | Duration of transmit path open from this peer to the voice gateway for this call. |
VADEnable | Whether voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmission from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value. |
show call history voice
show dial-peer voice
show num-exp
show voice port
To display the call history table, use the show call history voice privileged EXEC command.
show call history voice [last number | brief]
last number | (Optional) Displays the last calls connected, where the number of calls displayed is defined by the argument number. Valid values are from 1 to 2147483647. |
brief | (Optional) Displays a truncated version of the call history table. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
The call history table contains a listing of all calls connected through this router in descending time order since Voice over IP was enabled. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the argument number.
The following is sample output from the show call history voice command:
router# show call history voice brief
<ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>
tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms delay:<last>/<min>/<max>ms <codec>
Telephony <int>: tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
234 : 158305740hs.1280 +241 +9199 pid:0 Answer +3...
tx:3804/76080 rx:1358/27160 10 (normal call clearing.)
IP 172.24.96.200:16468 rtt:33ms pl:25990/0ms delay:30/30/70ms g729r8
234 : 158305745hs.1281 +236 +9195 pid:6 Originate +68888
tx:1358/27160 rx:3804/76080 10 (normal call clearing.)
Telephony 0:D:22: tx:91850/76080/0ms g729r8 noise:-84dBm acom:20dBm
235 : 158344850hs.1282 +230 +28773 pid:0 Answer +3...
tx:11063/221260 rx:4604/92080 10 (normal call clearing.)
IP 172.24.96.200:16474 rtt:41ms pl:88260/290ms delay:40/30/130ms g729r8
235 : 158344856hs.1283 +224 +28769 pid:6 Originate +68888
tx:4604/92080 rx:11063/221260 10 (normal call clearing.)
Telephony 0:D:22: tx:287590/221280/0ms g729r8 noise:-75dBm acom:20dBm
Table 8 provides an alphabetical listing of the possible fields for the show call history voice command and a description of each field.
| Field | Description |
|---|---|
ACOMLevel | Average ACOM level for this call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for a particular call. |
CallOrigin | Call origin: answer or originate. |
CoderTypeRate | Negotiated coder rate. This value specifies the transmit rate of voice/fax compression to its associated call leg for this call. |
ConnectionID | Global call identifier for the gateway call. |
ConnectTime | Time this call was connected. |
DisconnectCause | Description explaining why this call was disconnected. |
DisconnectText | Descriptive text explaining the disconnect reason. |
DisconnectTime | Time this call was disconnected. |
FaxDuration | Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value. |
GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received in time from the voice gateway for this call. |
GapFillWithRedundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because voice data was lost or not received in time from the voice gateway for this call. |
GapFillWithSilence | Duration of voice signal replaced with silence because the voice data was lost or not received in time for this call. |
GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during the voice call. |
Index | Dial peer identification number. |
InfoType | Information type for this call. |
LogicalIfIndex | Index number of the logical voice port for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the voice call. |
NoiseLevel | Average noise level for this call. |
OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
PeerAddress | Destination pattern or number associated with this peer. |
PeerId | ID value of the peer entry table to which this call was made. |
PeerIfIndex | Index number of the logical interface through which this call was made. For ISDN media, this would be the index number of the B channel used for this call. |
PeerSubAddress | Subaddress to which this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the Decoder Delay during this voice call. |
ReceivePackets | Number of packets received by this peer during this call. |
RemoteIPAddress | Remote system IP address for this call. |
RemoteUDPPort | Remote system UDP listener port to which voice packets are transmitted. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone for this call. |
SelectedQoS | Selected RSVP QoS for this call. |
Session Protocol | Session protocol to be used for an Internet call between the local and remote router via the IP backbone. |
Session Target | Session target of the peer used for thid call. |
SetUpTime | Value of the system UpTime when the call associated with this entry was started. |
TransmitBytes | Number of bytes transmitted by this peer during this call. |
TransmitPackets | Number of packets transmitted by this peer during this call. |
TxDuration | Duration of the transmit path open from this peer to the voice gateway for this call. |
VADEnable | Whether voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmitted from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration by the TxDuration value. |
show call active voice
show dial-peer voice
show num-exp
show voice port
modem | Specifies CSM call statistics for modems. |
voice | Specifies CSM call statistics for DSP channels. |
slot/port | (Optional) Specifies the location (and thereby the identity) of a specific modem. |
modem-group-number | (Optional) Displays configuration for the dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767. |
slot/dspm/dsp/dsp-channel | (Optional) Identifies the location of a particular DSP channel. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3 NA.
Use the show csm modem command to display the CSM call statistic information for a specific modem, for a group of modems, or for all modems. If a slot/port is specified, then CSM call statistics are displayed for the specified modem. If the modem-group-number keyword is specified, the CSM call statistics for all of the modems associated with that modem group are displayed. If no keyword is specified, CSM call statistics for all modems on the AS5300 are displayed.
Use the show csm voice command to display CSM statistics for a particular DSP channel. If the slot/dspm/dsp/dsp-channel is specified, the CSM call statistics for calls using the identified DSP channel will be displayed. If no argument is specified, all CSM call statistics for all DSP channels will be displayed.
The following is sample output from the show csm command:
suchan# show csm voice 2/4/4/0 slot 2, dspm 4, dsp 4, dsp channel 0, slot 2, port 56, tone, device_status(0x0002): VDEV_STATUS_ACTIVE_CALL. csm_state(0x0406)=CSM_OC6_CONNECTED, csm_event_proc=0x600E2678, current call thru PRI line invalid_event_count=0, wdt_timeout_count=0 wdt_timestamp_started is not activated wait_for_dialing:False, wait_for_bchan:False pri_chnl=TDM_PRI_STREAM(s0, u0, c22), tdm_chnl=TDM_DSP_STREAM(s2, c27) dchan_idb_start_index=0, dchan_idb_index=0, call_id=0xA003, bchan_num=22 csm_event=CSM_EVENT_ISDN_CONNECTED, cause=0x0000 ring_no_answer=0, ic_failure=0, ic_complete=0 dial_failure=0, oc_failure=0, oc_complete=3 oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0 remote_link_disc=0, stat_busyout=0 oobp_failure=0 call_duration_started=00:06:53, call_duration_ended=00:00:00, total_call_duration=00:00:44 The calling party phone number = 408 The called party phone number = 5271086 total_free_rbs_timeslot = 0, total_busy_rbs_timeslot = 0, total_dynamic_busy_rbs_timeslot = 0, total_static_busy_rbs_timeslot = 0, total_sw56_rbs_timeslot = 0, total_sw56_rbs_static_bo_ts = 0, total_free_isdn_channels = 21, total_busy_isdn_channels = 0,total_auto_busy_isdn_channels = 0, min_free_device_threshold = 0
Table 9 explains the fields contained in both of these examples.
| Field | Description |
|---|---|
slot | Indicates the slot where the VFC resides. |
dspm/dsp/dsp channel | Indicates which DSP channel is engaged in this call. |
dsp | Indicates the DSP through which this call is established. |
slot/port | This is the logical port number for the device. This is equivalent to the DSP channel number. The port number is derived from: |
tone | Indicates which signalling tone is being used (DTMF, MF, R2). This only applies to CAS calls. Possible values are:
|
device_status | The status of the device. Possible values are:
|
csm_state | CSM call state of the current call (PRI line) associated with this device. Possible values are:
|
csm_state: invalid_event_count= | Number of invalid events received by the CSM state machine. |
wdt_timeout_count= | Number of times the watchdog timer is activated for this call. |
wdt_timestamp_started | Indicates whether the watchdog timer is activated for this call. |
wait_for_dialing:
| Indicates whether this (outgoing) call is waiting for a free digit collector to become available to dial out the outgoing digits. |
wait_for_bchan: | Indicates whether this (outgoing) call is waiting for a B channel to send the call out on. |
pri_chnl= | Indicates which type of TDM stream is used for the PRI connection. For PRI and CAS calls, it will always be TDM_PRI_STREAM. |
tdm_chnl= | Indicates which type of TDM stream is used for the connection to the device used to process this call. In the case of a VoIP call, this will always be set to TDM_DSP_STREAM. |
dchan_idb_start_index= | First index to use when searching for the next IDB of a free D channel. |
dchan_idb_index= | Index of the currently available IDB of a free D channel. |
csm_event= | Event just passed to the CSM state machine. |
cause | Event cause. |
ring_no_answer= | Number of times call failed because there was no response. |
ic_failure= | Number of failed incoming calls. |
ic_complete= | Number of successful incoming calls. |
dial_failure= | Number of times the connection failed because there was no dial tone. |
oc_failure= | Number of failed outgoing calls. |
oc_complete= | Number of successful outgoing calls. |
oc_busy= | Number of outgoing calls where the connection failed because there was a busy signal. |
oc_no_dial_tone= | Number of outgoing calls where the connection failed because there was no dial tone. |
oc_dial_timeout= | Number of outgoing calls where the connection failed because the timeout value was exceeded. |
call_duration_started= | Indicates the start of this call. |
call_duration_ended= | Indicates the end of this call. |
total_call_duration= | Indicates the duration of this call. |
The calling party phone number = | Calling party number as given to CSM by ISDN. |
The called party phone number = | Called party number as given to CSM by ISDN. |
total_free_rbs_timeslot = | Total number of free RBS (CAS) timeslots available for the whole system. |
total_busy_rbs_timeslot = | Total number of RBS (CAS) timeslots that have been busied out. This includes both dynamically and statically busied out RBS timeslots. |
total_dynamic_busy_rbs_timeslot = | Total number of RBS (CAS) timeslots that have been dynamically busied out. |
total_static_busy_rbs_timeslot = | Total number of RBS (CAS) timeslots that have been statically busied out (that is, they are busied out using the CLI command) |
total_free_isdn_channels = | Total number of free ISDN channels. |
total_busy_isdn_channels = | Total number of busy ISDN channels. |
total_auto_busy_isdn_channels = | Total number of ISDN channels that are automatically busied out. |
show call active voice
show call-history voice
show num-exp
show voice port
number | (Optional) Displays configuration for the dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show dial-peer voice privileged EXEC command to display the configuration for all VoIP and POTS dial peers configured for the router. To show configuration information for only one specific dial peer, use the number argument to identify that dial peer.
The following is sample output from the show dial-peer voice command for a POTS dial peer:
router# show dial-peer voice 1
VoiceEncapPeer1
tag = 1, dest-pat = \Q+14085291000',
answer-address = \Q',
group = 0, Admin state is up, Operation state is down
Permission is Both,
type = pots, prefix = \Q',
session-target = \Q', voice-port =
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
The following is sample output from the show dial-peer voice command for a VoIP dial peer:
router# show dial-peer voice 10
VoiceOverIpPeer10
tag = 10, dest-pat = \Q',
incall-number = \Q+14087',
group = 0, Admin state is up, Operation state is down
Permission is Answer,
type = voip, session-target = \Q',
sess-proto = cisco, req-qos = bestEffort,
acc-qos = bestEffort,
fax-rate = voice, codec = g729r8,
Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled,
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
Table 10 explains the fields contained in both of these displays.
| Field | Description |
|---|---|
Accepted Calls | Number of calls from this peer accepted since system startup. |
acc-qos | Lowest acceptable quality of service configured for calls for this peer. |
Admin state | Administrative state of this peer. |
Charged Units | Total number of charging units applying to this peer since system startup. The unit of measure is in hundredths of seconds. |
codec | Default voice coder rate of speech for this peer. |
Connect Time | Accumulated connect time to the peer since system startup for both incoming and outgoing calls. The unit of measure is in hundredths of seconds. |
dest-pat | Destination pattern (telephone number) for this peer. |
Expect factor | User-requested Expectation Factor of voice quality for calls via this peer. |
fax-rate | Fax transmission rate configured for this peer. |
Failed Calls | Number of failed call attempts to this peer since system startup. |
group | Group number associated with this peer. |
Icpif | Configured Calculated Planning Impairment Factor (ICPIF) value for calls sent by a dial peer. |
incall-number | Full E.164 telephone number to be used to identify this dial peer. |
Last Disconnect Cause | Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface. |
Last Disconnect Text | ASCII text describing the reason for the last call termination. |
Last Setup Time | Value of the System Up Time when the last call to this peer was started. |
Operation state | Operational state of this peer. |
Permission | Configured permission level for this peer. |
Poor QOV Trap | Whether Poor Quality of Voice trap messages have been enabled or disabled. |
Refused Calls | Number of calls from this peer refused since system startup. |
req-qos | Configured requested quality of service for calls for this dial peer. |
session-target | Session target of this peer. |
sess-proto | Session protocol to be used for Internet calls between local and remote routers via the IP backbone. |
Successful Calls | Number of completed calls to this peer. |
tag | Unique dial peer ID number. |
VAD | Whether voice activation detection (VAD) is enabled for this dial peer. |
show call active voice
show call-history voice
show num-exp
show voice port
controller number | Specifies the T1 or E1 controller. |
cas-group:number | Specifies the CAS group number. |
:D | Indicates the D channel associated with ISDN PRI. |
dial-string | Specifies a particular destination pattern (telephone number). |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Occasionally, an incoming call cannot be matched to a dial peer in the dial peer database. One reason an incoming call cannot be matched to a dial peer is that the specified destination cannot be reached via the voice interface through which the incoming call came. Use the show dialplan incall number command as a troubleshooting method to resolve the call destination by pairing voice ports and telephone numbers together until there is a match.
The following example tests whether the telephone extension 57681 can be reached through voice port 0:D:
show dialplan incall 0:D number 57681
show dialplan number
dial-string | Specifies a particular destination pattern (telephone number). |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
The following example displays the dial peer associated with the destination pattern of 54567:
router# show dialplan number 51234
Macro Exp.: 14085551234
VoiceOverIpPeer1004
tag = 1004, destination-pattern = \Q+1408555....',
answer-address = \Q',
group = 1004, Admin state is up, Operation state is up
type = voip, session-target = \Qipv4:1.13.24.0',
ip precedence: 0 UDP checksum = disabled
session-protocol = cisco, req-qos = best-effort,
acc-qos = best-effort,
fax-rate = voice, codec = g729r8,
Expect factor = 10, Icpif = 30,
VAD = enabled, Poor QOV Trap = disabled
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
Matched: +14085551234 Digits: 7
Target: ipv4:172.13.24.0
Table 11 explains the fields contained in this example.
| Field | Description |
|---|---|
Macro Exp. | Expected destination pattern for this dial peer. |
VoiceOverIpPeer | Identifies the dial peer associated with the destination pattern entered. |
tag | Unique dial peer identifying number |
destination-pattern | Destination pattern (telephone number) configured for this dial peer |
answer-address | Answer address configured for this dial peer. |
Admin state | Describes the administrative state of this dial peer. |
Operation state | Describes the operational state of the dial peer. |
type | Type of dial peer (POTS or VoIP). |
session-target | Displays the configures session target (IP address or host name) for this dial peer. |
ip precedence | Displays the numeric value for the IP Precedence configured for this dial peer. |
UDP checksum | Indicates the status of the UDP checksum feature. |
session-protocol | Session protocol to be used for Internet calls between local and remote router via the IP backbone. |
req-qos | Configured requested quality of service for calls for this dial peer. |
acc-qos | Configures acceptable quality of service for calls for this dial peer. |
fax-rate | Configured facsimile transmission speed for with this dial peer. |
codec | CODEC type configured for this dial peer. |
Expect factor | Configured value at which the system will generate an SMTP message alerting that the voice quality has dropped. |
Icpif | Configured Calculated Planning Impairment Factor (ICPIF) value for calls sent by a dial peer. |
VAD | Whether or not voice activation detection (VAD) is enabled for this dial peer. |
Poor QOV Trap | Whether Poor Quality of Voice trap messages have been enabled or disabled. |
Connect Time | Unit of measure indicating the call connection time associated with this dial peer. |
Charged Units | Number of call units charged to this dial peer. |
Successful Calls | Number of completed calls to this peer since system startup. |
Failed Calls | Number of uncompleted (failed) calls to this peer since system startup. |
Accepted Calls | Number of calls from this peer accepted since system startup. |
Refused Calls | Number of calls from this peer refused since system startup. |
Last Disconnect Cause | Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface. |
Last Disconnect Text | ASCII text describing the reason for the last call termination. |
Last Setup Time | Value of the System Up Time when the last call to this peer was started. |
Matched | Destination pattern matched for this dial peer. |
Target | Matched session target (IP address or host name) for this dial peer. |
show dialplan incall number
To show the number expansions configured, use the show num-exp privileged EXEC command.
show num-exp [dialed-number]
dialed-number | (Optional) Displays number expansion for the specified dialed number. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show num-exp privileged EXEC command to display all of the number expansions configured for this router. To display number expansion for only one number, specify that number by using the dialed-number argument.
The following is sample output from the show num-exp command:
router# show num-exp Dest Digit Pattern = '0...' Translation = '+14085270...' Dest Digit Pattern = '1...' Translation = '+14085271...' Dest Digit Pattern = '3..' Translation = '+140852703..' Dest Digit Pattern = '4..' Translation = '+140852804..' Dest Digit Pattern = '5..' Translation = '+140852805..' Dest Digit Pattern = '6....' Translation = '+1408526....' Dest Digit Pattern = '7....' Translation = '+1408527....' Dest Digit Pattern = '8...' Translation = '+14085288...'
Table 12 explains the fields in the sample output.
| Field | Description |
|---|---|
Dest Digit Pattern | Index number identifying the destination telephone number digit pattern. |
Translation | Expanded destination telephone number digit pattern. |
show call active voice
show call history voice
show dial-peer voice
show voice port
slot | Identifies the slot where the VFC is installed. Valid entries are from 0 to 2. |
User EXEC
This command first appeared in Cisco IOS Release 11.3 NA.
To identify the specific VFC, enter the number of the slot on the chassis where the VFC resides using the slot argument.
The following is sample output from the show vfc cap-list command:
router> show vfc 1 cap-list Capability List for VFC in slot 1: 1. fax-vfc-l.0.1.bin 2. bas-vfc-l.0.1.bin 3. cdc-g729-l.0.1.bin 4. cdc-g711-l.0.1.bin 5. cdc-g726-l.0.1.bin 6. cdc-g728-l.0.1.bin 7. cdc-gsmfr-l.0.1.bin
The first line in this output is a general description, stating that this is the capability list for the VFC residing in slot 1. Below this is a numbered list, each line of which identifies one currently installed in-service file.
show vfc default-file
show vfc directory
show vfc version
slot | Identifies the slot where the VFC is installed. Valid entries are from 0 to 2. |
User EXEC
This command first appeared in Cisco IOS Release 11.3 NA.
Use the show vfc default-file user EXEC command to display a list of all default files for a particular voice feature card. To identify the specific VFC, enter the number of the slot on the chassis where the VFC resides using the slot argument.
The following is sample output from the show vfc default-file command:
router> show vfc 1 default-file Default List for VFC in slot 1: 1. btl-vfc-l.0.13.0.bin 2. cor-vfc-l.0.1.bin 3. bas-vfc-l.0.1.bin 4. cdc-g729-l.0.1.bin 5. fax-vfc-l.0.1.bin 6. jbc-vfc-l.0.13.0.bin
The first line in this output is a general description, stating that this is the default list for the VFC residing in slot 1. Below this is a numbered list, each line of which identifies one default file.
show vfc cap-list
show vfc directory
show vfc version
To show the list of all files residing on this VFC, use the show vfc directory user EXEC command.
show vfc slot directory
slot | Identifies the slot where the VFC is installed. Valid entries are from 0 to 2. |
User EXEC
This command first appeared in Cisco IOS Release 11.3 NA.
Use the show vfc directory user EXEC command to display a list of all of the files currently stored in Flash memory for a particular VFC. To identify the specific VFC, enter the number of the slot on the chassis where the VFC resides using the slot argument.
The following is sample output from the show vfc directory command:
router> show vfc 1 directory
Files in slot 1 VFC flash:
File Name Size (Bytes)
1 . vcw-vfc-mz.gsm.VCW 292628
2 . btl-vfc-l.0.13.0.bin 4174
3 . cor-vfc-l.0.1.bin 54560
4 . jbc-vfc-l.0.13.0.bin 16760
5 . fax-vfc-l.0.1.bin 64290
6 . bas-vfc-l.0.1.bin 54452
7 . cdc-g711-l.0.1.bin 190
8 . cdc-g729-l.0.1.bin 21002
9 . cdc-g726-l.0.1.bin 190
10. cdc-g728-l.0.1.bin 22270
11. cdc-gsmfr-l.0.1.bin 190
Table 13 explains the fields in the sample output.
| Field | Description |
|---|---|
File Name | Name of the file stored in Flash memory. |
Size (Bytes) | Size of the file in bytes. |
show vfc cap-list
show vfc default-file
show vfc version
To show the version of the software residing on this VFC, use the show vfc version user EXEC command.
show vfc slot version {dspware | vcware}
slot | Identifies the slot where the VFC is installed. Valid values are 0, 1, and 2. |
dspware | Defines which DSPWare software to display. |
vcware | Defines which VCWare software to display. |
User EXEC
This command first appeared in Cisco IOS Release 11.3 NA.
Use the show vfc version user EXEC command to display the version of the software (either running on DSP or VFC) currently installed in Flash memory on the VFC.
The following is sample output from the show vfc version command:
router> show vfc 0 version dspware Version of Dspware in VFC slot 0 is 0.10
The output from this command is a simple declarative sentence stating the version number for the selected type of software (in this example, DSPWare) for the VFC residing in the selected slot number (in this example, slot 0).
show vfc cap-list
show vfc default-file
show vfc directory
controller number | Specifies the T1 or E1 controller. |
:D | Indicates the D channel associated with ISDN PRI. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(2)NA.
Use the show voice port privileged EXEC command to display configuration and voice interface card-specific information about a specific port.
The following is sample output from the show voice port command:
router#show voice port 0:DISDN 0:DType of VoicePort is ISDNOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is ""Noise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 16 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sRegion Tone is set for US
Table 14 explains the fields in the sample output.
| Field | Description |
|---|---|
Type of VoicePort | Indicates the voice port type. |
Operation State | Operational state of this voice port. |
Administrative State | Administrative state of this voice port. |
Description | Descriptive text attached to this voice port. |
Noise Regeneration | Whether background noise should be played to fill silent gaps if VAD is activated. |
Non Linear Processing | Whether non linear processing is enabled for this port. |
Music-On-Hold Threshold | Configured Music-On-Hold Threshold value for this interface. |
In Gain | Amount of gain inserted at the receiver side of this interface. |
Out Attenuation | Amount of attenuation inserted at the transmit side of this interface. |
Echo Cancellation | Whether echo cancellation is enabled for this port. |
Echo Cancel Coverage | The configured value for the Echo Canceller for this port. |
Connection Mode | Connection mode of this interface. |
Connection Number | Full E.164 telephone number used to establish a connection with the trunk or PLAR mode. |
Initial Time Out | Amount of time the system waits for an initial input digit from the caller. |
Interdigit Time Out | Amount of time the system waits for a subsequent input digit from the caller. |
Regional Tone | Configured regional tone for this interface. |
show call active voice
show call history voice
show dial-peer voice
show num-exp
This command has no arguments or keywords.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
When a dial peer is shut down, you cannot initiate calls to that peer. This command applies to both VoIP and POTS peers.
The following example changes the administrative state of voice telephony dial peer 10 to down:
configure terminal dial-peer voice 10 pots shutdown
This command has no arguments or keywords.
Disabled
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the snmp enable peer-trap poor qov command to generate poor quality of voice notifications for applicable calls associated with this dial peer. If you have an SNMP manager that will use SNMP messages when voice quality drops, you might want to enable this command. Otherwise, you should disable this command to reduce unnecessary network traffic.
This command applies only to VoIP peers.
The following example enables poor quality of voice notifications for calls associated with VoIP dial peer 10:
dial-peer voice 10 voip snmp enable peer-trap poor-qov
snmp-server enable trap voice poor-qov
snmp trap link-status
To enable the router to send SNMP traps, use the snmp-server enable traps global configuration command. Use the no form of this command to disable SNMP traps.
snmp-server enable traps [trap-type] [trap-option]
trap-type | (Optional) Type of trap to enable. If no type is specified, all traps are sent (including the envmon and repeater traps). The trap type can be one of the following keywords: · bgp---Sends Border Gateway Protocol (BGP) state change traps. · config---Sends configuration traps. · entity---Sends Entity MIB modification traps. · envmon---Sends Cisco enterprise-specific environmental monitor traps when an environmental threshold is exceeded. When the envmon keyword is used, you can specify a trap-option value. · frame-relay---Sends Frame Relay traps. · isdn---Sends Integrated Services Digital Network (ISDN) traps. When the isdn keyword is used on Cisco 1600 series routers, you can specify a trap-option value. · repeater---Sends Ethernet hub repeater traps. When the repeater keyword is selected, you can specify a trap-option value. · rtr---Sends Response Time Reporter (RTR) traps. · snmp---Sends Simple Network Management Protocol (SNMP) traps. When the snmp keyword is used, you can specify a trap-option value. · syslog---Sends error message traps (Cisco Syslog MIB). Specify the level of messages to be sent with the logging history level command. · voice---Sends SNMP poor quality of voice traps, when used with the qov trap-option. |
trap-option | (Optional) When the envmon keyword is used, you can enable a specific environmental trap type, or accept all trap types from the environmental monitor system. If no option is specified, all environmental types are enabled. The option can be one or more of the following keywords: voltage, shutdown, supply, fan, and temperature. When the isdn keyword is used on Cisco 1600 series routers, you can specify the call-information keyword to enable an SNMP ISDN call information trap for the ISDN MIB subsystem, or you can specify the isdnu-interface keyword to enable an SNMP ISDN U interface trap for the ISDN U interface MIB subsystem. When the repeater keyword is used, you can specify the repeater option. If no option is specified, all repeater types are enabled. The option can be one or more of the following keywords: · health---Enables IETF Repeater Hub MIB (RFC 1516) health trap. · reset---Enables IETF Repeater Hub MIB (RFC 1516) reset trap. When the snmp keyword is used, you can specify the authentication option to enable SNMP Authentication Failure traps. (The snmp-server enable traps snmp authentication command replaces the snmp-server trap-authentication command.) If no option is specified, all SNMP traps are enabled. When the voice keyword is used, you can enable SNMP poor quality of voice traps by using the qov option. |
This command is disabled by default. No traps are enabled.
If you enter this command with no keywords, the default is to enable all trap types.
Some trap types cannot be controlled with this command. These traps are either always enabled or enabled by some other means. For example, the linkUpDown messages are disabled by the no snmp trap link-status command.
Global configuration
This command first appeared in Cisco IOS Release 11.1.
This command is useful for disabling traps that are generating a large amount of uninteresting or useless noise.
If you do not enter an snmp-server enable traps command, no traps controlled by this command are sent. In order to configure the router to send these SNMP traps, you must enter at least one snmp-server enable traps command. If you enter the command with no keywords, all trap types are enabled. If you enter the command with a keyword, only the trap type related to that keyword is enabled. To enable multiple types of traps, you must issue a separate snmp-server enable traps command for each trap type and option.
The snmp-server enable traps command is used in conjunction with the snmp-server host command. Use the snmp-server host command to specify which host or hosts receive SNMP traps. To send traps, you must configure at least one snmp-server host command.
For a host to receive a trap controlled by this command, both the snmp-server enable traps command and the snmp-server host command for that host must be enabled. If the trap type is not controlled by this command, only the appropriate snmp-server host command must be enabled.
The trap types used in this command all have an associated MIB object that allows them to be globally enabled or disabled. Not all of the trap types available in the snmp-server host command have notificationEnable MIB objects, so some of these cannot be controlled using the snmp-server enable traps command.
The following example enables the router to send SNMP poor quality of voice traps:
configure terminal
snmp-server enable trap voice poor-qov
The following example enables the router to send all traps to the myhost.cisco.com host using the public community string:
snmp-server enable traps snmp-server host myhost.cisco.com public
The following example enables the router to send Frame Relay and environmental monitor traps to the myhost.cisco.com host using the public community string:
snmp-server enable traps frame-relay snmp-server enable traps envmon temperature snmp-server host myhost.cisco.com public
The following example will not send traps to any host. The BGP traps are enabled for all hosts, but the only traps enabled to be sent to a host are ISDN traps.
snmp-server enable traps bgp snmp-server host bob public isdn
snmp enable peer-trap peer-qov
snmp-server host
snmp-server trap-source
snmp trap illegal-address
snmp trap link-status
This command contains no arguments or keywords.
Enabled
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the snmp trap link-status command to enable SNMP trap messages (linkup and linkdown) to be generated whenever this voice port is brought online or offline.
If you are managing the equipment with an SNMP manager (such as Maestro), this command should be enabled. Enabling link-status messages will allow the SNMP manager to learn of a status change without polling the equipment. If you are not using an SNMP manager, this command should be disabled to avoid unnecessary network traffic.
The following example enables SNMP trap messages for voice port 2/1/0:
voice-port 2/1/0 snmp trap link-stat
smnp enable peer-trap poor-qov
snmp-server enable trap poor-qov
seconds | Specifies the initial timeout duration in seconds. Valid entries are any integer from 0 to 120. |
The default value is 10 seconds.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the timeouts initial command to specify the number of seconds the system will wait for the caller to input the first digit of the dialed digits. The timeouts initial timer is activated when the call is accepted and deactivated when the caller inputs the first digit. If the configured timeout value is exceeded, the caller is notified by the appropriate tone, and the call is terminated.
To disable the timeouts initial timer, set the seconds value to 0.
The following example sets the initial digit timeout value to 15 seconds:
voice-port 1/0/0 timeouts initial 15
timeouts interdigit
timing
seconds | Specifies the interdigit timeout duration in seconds. Valid entries are any integer from 0 to 120. |
The default value is 10 seconds.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the timeouts interdigit command to specify the number of seconds the system will wait after the caller has input the initial digit for the caller to input a subsequent digit of the dialed digits. The timeouts interdigit timer is activated each time the caller inputs a digit until the destination address is identified. If the configured timeout value is exceeded before the destination address is identified, the caller is notified by the appropriate tone, and the call is terminated.
To disable the timeouts interdigit timer, set the seconds value to 0.
The following example sets the interdigit timeout value for 15 seconds:
voice-port 1/0/0 timeouts interdigit 15
timeouts initial
timing
slot | Specifies the slot in the Cisco AS5300 where the VFC resides. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(2)NA.
VFCs come with a single bundled image, VCWare, stored in VFC Flash memory. Use the unbundle vfc command to unbundle this bundled image into separate files, which are then written to Flash memory. When VCWare is unbundled, it automatically adds DSPWare to Flash memory, creates both the capability and default file lists, and populates these lists with the default files for that version of VCWare. The default file list includes the files that will be used to boot up the system. The capability list defines the available voice CODECS for H.323 capability negotiation. These files are used during initial card configuration and for subsequent firmware upgrades.
The following example unbundles files from the VFC located in slot 1 into separate files in VFC Flash memory:
unbundle vfc 1
copy flash vfc
copy tftp vfc
This command has no arguments or keywords.
Enabled
Dial-peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the vad command to enable voice activity detection. With VAD, silence is not transmitted over the network, only audible speech. If you enable VAD, the sound quality will be slightly degraded, but the connection will monopolize much less bandwidth. If you use the no form of this command, VAD is disabled and voice data is continuously transmitted to the IP backbone.
This command applies only to VoIP peers.
The following example enables VAD:
dial-peer voice 10 voip vad
comfort-noise
To enter the voice-port configuration mode, use the voice-port global configuration command.
voice-port controller number:D
controller number | Specifies the T1 or E1 controller. |
:D | Indicates the D channel associated with ISDN PRI. |
Global configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the voice-port global configuration command to switch to the voice-port configuration mode from the global configuration mode. Use the exit command to exit the voice-port configuration mode and return to the global configuration mode.
The following example accesses the voice-port configuration mode:
configure terminal voice-port 1:D
dial-peer
Posted: Wed Jun 23 18:19:49 PDT 1999
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