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vad (dial peer) through voice-class sip encap clear-channel
- vad (dial peer)
- vad (SPA-DSP)
- vbd-playout-delay
- vbr-rt
- vcci
- video codec (dial peer)
- video codec (voice class)
- video screening
- violation
- violation (media profile)
- vmwi
- vofr
- voice
- voicecap configure
- voicecap entry
- voice call capacity mir
- voice call capacity reporting
- voice call capacity stw
- voice call capacity timer interval
- voice call convert-discpi-to-prog
- voice call csr data-points
- voice call csr recording interval
- voice call csr reporting interval
- voice call debug
- voice call disc-pi-off
- voice call rate monitor
- voice call send-alert
- voice call trap deviation
- voice call trigger hwm
- voice call trigger lwm
- voice call trigger percent-change
- voice-card
- voice cause-code
- voice class aaa
- voice class busyout
- voice class called number
- voice class cause-code
- voice class codec
- voice class custom-cptone
- voice class dscp-profile
- voice class dualtone
- voice class dualtone-detect-params
- voice class e164-pattern-map
- voice class e164-pattern-map load
- voice class h323
- voice class media
- voice class permanent
- voice class resource-group
- voice class sip-copylist
- voice class sip-profiles
- voice class tone-signal
- voice class uri
- voice class uri sip preference
- voice-class aaa (dial peer)
- voice-class called-number (dial peer)
- voice-class called-number-pool
- voice-class codec (dial peer)
- voice-class h323 (dial peer)
- voice-class permanent (dial-peer)
- voice-class permanent (voice-port)
- voice-class sip anat
- voice pcm capture
- voice-class sip asserted-id
- voice-class sip associate registered-number
- voice-class sip asymmetric payload
- voice-class sip authenticate redirecting-number
- voice-class sip bind
- voice-class sip block
- voice-class sip call-route
- voice-class sip calltype-video
- voice-class sip copy-list
- voice-class sip e911
- voice-class sip encap clear-channel
vad (dial peer)
To enable voice activity detection (VAD) for calls using a specific dial peer, use the vad command in dial-peer configuration mode. To disable VAD, use the no form of this command.
Usage Guidelines
Use thiscommand to enable voice activity detection. With VAD, voice data packets fall into three categories: speech, silence, and unknown. Speech and unknown packets are sent over the network; silence packets are discarded. The sound quality is slightly degraded with VAD, but the connection monopolizes much less bandwidth. If you use the no form of this command, VAD is disabled and voice data is continuously sent to the IP backbone. When configuring voice gateways to handle fax calls, VAD should be disabled at both ends of the IP network because it can interfere with the successful reception of fax traffic.
When the aggressive keyword is used, the VAD noise threshold is reduced from -78 to -62 dBm. Noise that falls below the -62 dBm threshold is considered to be silence and is not sent over the network. Additionally, unknown packets are considered to be silence and are discarded.
Examples
The following example enables VAD for a Voice over IP (VoIP) dial peer, starting from global configuration mode:
dial-peer voice 200 voip vadRelated Commands
Command
Description
comfort-noise
Generates background noise to fill silent gaps during calls if VAD is activated.
dial-peer voice
Enters dial-peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
vad (voice-port)
Enables VAD for the calls using a particular voice port.
vad (SPA-DSP)
To enable or disable voice activity detection (vad) settings configured locally irrespective of the external vad settings, use the vadcommand in config dspfarm profile mode.
Usage Guidelines
Use this command to enable voice activity detection locally irrespective of external VAD settings. With VAD, voice data packets fall into three categories: speech, silence, and unknown. Speech and unknown packets are sent over the network; silence packets are discarded. The sound quality is slightly degraded with VAD, but the connection monopolizes much less bandwidth. If you disable VAD, voice data is continuously sent to the IP backbone.
Examples
The following example enables VAD and overrides external vad settings with local vad settings:
Router(config)# dspfarm profile 1 Router(config-dspfarm-profile)# vad on override Router(config-dspfarm-profile)# do show running-config !!! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 588 associate application SBC vad on override !The following example disables local vad settings and overrides external vad setting configuration:
Router(config)# dspfarm profile 1 Router(config-dspfarm-profile)# vad off override Router(config-dspfarm-profile)# do show running-config !!! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 588 associate application SBC vad off override !Related Commands
Command
Description
dsp services dspfarm
Enables the DSP-farm services.
dspfarm profile
Enters the DSP farm profile configuration mode, and defines a profile for the DSP farm services.
show dspfarm (SPA-DSP)
Displays DSP farm service information, such as operational status and DSP resource allocation for transcoding.
vbd-playout-delay
To configure the voice-band-detection playout-delay buffer on a Cisco router, use the vbd-playout-delay command invoice service session configuration mode. To disable the buffer, use the no form of this command.
vbd-playout-delay { maximum milliseconds | minimum milliseconds | mode { fixed [no-timestamps] | passthrough } | nominal milliseconds }
no vbd-playout-delay
Syntax Description
maximum
Sets the maximum playout buffer delay, in milliseconds (ms). Range: 40 to 1000. Default: 1000.
milliseconds
Delay time, in milliseconds (ms).
minimum
Sets the minimum playout buffer delay, in ms. Range: 10 to 40. Default: 40.
mode
Configures voice-band-detection playout buffer adaptation mode.
fixed
Sets the jitter buffer to a constant delay.
no-timestamps
(Optional) Fixes the jitter buffer at a constant delay without time stamps.
passthrough
Sets the jitter buffer passthrough mode for clock compensation.
nominal
Sets the nominal playout buffer delay, in ms. Range:10 to 1000. Default: 60.
Command History
Usage Guidelines
Use thiscommand to set the playout jitter buffer. When a voice band is detected, the call uses the G.711 codec, and the playout delay values that you set are picked up. The original voice-call parameters are restored after the fax or modem call is completed. The no-timestamps keyword sets the jitter buffer at a constant delay without reading time stamps.
NoteThe passthough keyword is a special mode used to handle clock drifting properly. We recommend this keyword only when instructed by your Cisco representative.
Examples
The following example configures ATM adaptation layer 2 (AAL2) voice-band-detection playout-delay adaptation mode and sets the mode to fixed:
voice service voatm session protocol aal2 vbd-playout-delay mode fixedThe following example configures AAL2 voice-band-detection playout-delay adaptation mode and sets the mode at a constant delay without timestamps:
voice service voatm session protocol aal2 vbd-playout-delay mode fixed no-timestampsThe following example sets the nominal AAL2 voice-band-detection playout-delay buffer to 12 ms:
voice service voatm session protocol aal2 vbd-playout-delay nominal 12The following example sets the AAL2 voice-band-detection playout-buffer delay to a maximum of 55 ms:
voice service voatm session protocol aal2 vbd-playout-delay maximum 55The following example sets the AAL2 voice-band-detection playout-buffer delay to a minimum of 22 ms:
voice service voatm session protocol aal2 vbd-playout-delay minimum 22The following sample output shows the vdb-playout-delay being verified in the running configuration output:
Router(conf-voi-serv-sess)#do show run | sec voice service voatm voice service voatm ! session protocol aal2 vbd-playout-delay minimum 22vbr-rt
To configure the real-time variable bit rate (VBR) for VoATM voice connections, use the vbr-rt command in the appropriate configuration mode. To disable VBR for voice connections, use the no form of this command.
Syntax Description
peak-rate
Peak information rate (PIR) for the voice connection, in kilobytes per second (kbps). If it does not exceed your carrier’s line rate, set it to the line rate. Range is from 56 to 10000.
average-rate
Average information rate (AIR) for the voice connection, in kbps.
burst
Burst size, in number of cells. Range is from 0 to 65536.
Command Modes
ATM Bundle-vc configuration for ATM VC bundle members
ATM PVP configuration for an ATM PVP
Interface-ATM-VC configuration for an ATM permanent virtual connection (PVC) or switched virtual circuit (SVC)
VC-class configuration for a virtual circuit (VC) classCommand History
Release
Modification
12.0
This command was introduced on the Cisco MC3810.
12.1(5)XM
This command was implemented on Cisco 3600 series routers and modified to support Simple Gateway Control Protocol (SGCP) and Media Gateway Control Protocol (MGCP).
12.2(2)T
This command was integrated into Cisco IOS Release 12.2(2)T.
12.2(11)T
This command was implemented on the Cisco AS5300 and Cisco AS5850.
Cisco IOS XE Release 2.3
This command was made available in ATM PVP configuration mode.
Usage Guidelines
This command configures traffic shaping between voice and data PVCs. Traffic shaping is required so that the carrier does not discard calls. To configure voice and data traffic shaping, you must configure the peak, average, and burst options for voice traffic. Configure the burst value if the PVC will carry bursty traffic. Peak, average, and burst values are needed so that the PVC can effectively handle the bandwidth for the number of voice calls.
Calculate the minimum peak, average, and burst values for the number of voice calls as follows:
Peak Value
Peak value = (2 x the maximum number of calls) x 16K = _______________
Average Value
Calculate according to the maximum number of calls that the PVC will carry times the bandwidth per call. The following formulas give you the average rate in kbps:
Average value = max calls x 128K = _______________
Average value = max calls x 85K = _______________
Average value = max calls x 85K = _______________
Average value = max calls x 85K = _______________
Average value = max calls x 43K = _______________
Average value = max calls x 43K = _______________
If voice activity detection (VAD) is enabled, bandwidth usage is reduced by as much as 12 percent with the maximum number of calls in progress. With fewer calls in progress, bandwidth savings are less.
Burst Value
Set the burst size as large as possible, and never less than the minimum burst size. Guidelines are as follows:
- Minimum burst size = 4 x number of voice calls = _______________
- Maximum burst size = maximum allowed by the carrier = _______________
When you configure data PVCs that will be traffic shaped with voice PVCs, use AAL5snap encapsulation and calculate the overhead as 1.13 times the voice rate.
vcci
To identify a permanent virtual circuit (PVC) to the call agent, use the vcci command in ATM virtual circuit (VC) configuration mode. To restore the default value, use the noform of this command.
Usage Guidelines
The pvc-identifier argumentis a unique 15-bit value for each PVC. The call agent sets up a call with the gateway by specifying the PVC using the pvc-identifier.
video codec (dial peer)
To assign a video codec to a VoIP dial peer, use the video codec command in dial peer configuration mode. To remove a video codec, use the no form of this command.
Usage Guidelines
Use this command to configure a video codec for a VoIP dial peer. If no video codec is configured, the default is transparent codec operation between the endpoints.
video codec (voice class)
To specify a video codec for a voice class, use the video codeccommand in voice class configuration mode. To remove the video codec, use the no form of this command.
video screening
To enable transcoding and transsizing between two call legs when configuring SIP, use the video screeningcommand in foice service SIP configuration mode. To disable transcoding and transsizing, use no form of this command.
Usage Guidelines
Use this command to enable conversion of video streams if there is a mismatch between two call legs.
violation
To specify the action that needs to be performed on any violation in the Differentiated Services Code Point (DSCP) policy, use the violation command in voice class configuration mode. To disable the configuration, use the no form of this command.
violation number action { disconnect | ignore } [ no-syslog ]
no violation number action { disconnect | ignore } [ no-syslog ]
Syntax Description
number
Number of violations after which the required action needs to be taken. The range is from 1 to 200000. The default value is 20.
action
Specifies that an action must be performed after the specified number of violations.
disconnect
Disconnects the call after the specified number of violations is exceeded.
ignore
Specifies that no action should be taken after the specified number of violations is exceeded.
no-syslog
(Optional) Specifies not to print messages to the system log when violations occur.
Usage Guidelines
You can use the violation command to specify the action that needs to be performed on any violation in the DSCP policy. A system log is created by default. You can configure the no-syslog keyword to disable the Cisco Unified Border Element (Cisco UBE) from generating system logs on DSCP policy violation.
Configure a high value for DSCP violations. If you configure a low value such as 5, action will be performed on the call after every five violations and system logs will be generated frequently.
The “100 - Invalid information element contents [Q.850]” message is displayed in the system log when a call is disconnected because of a DSCP policy violation. The cause for disconnecting the call is propagated only to the call leg causing the violation. For example, if the outgoing call leg of a Session Initiation Protocol (SIP)-to-SIP call violates the DSCP policy and the number of violations exceeds the configured number, this call is disconnected with the cause of 100 (Invalid information element contents [Q.850]) to the outgoing call leg and cause 16 (Normal Call Cleaning) to the incoming call leg.
violation (media profile)
To specify the action that needs to be performed on any violation in the media bandwidth policy, use the violation command in media profile configuration mode. To disable the configuration, use the no form of this command.
violation number action { disconnect | drop | ignore } [ no-syslog ]
no violation number action { disconnect | drop | ignore } [ no-syslog ]
Syntax Description
number
Number of violations after which the required action needs to be taken. The range is from 1 to 200000. The default value is 20.
action
Specifies that an action must be performed after the specified number of violations.
disconnect
Disconnects the call after the specified number of violations is exceeded.
drop
Drops the call after the specified number of violations is exceeded.
ignore
Specifies that no action should be taken after the specified number of violations is exceeded.
no-syslog
(Optional) Specifies not to print messages to the system log when violations occur.
Usage Guidelines
You can use the violation command to specify the action that needs to be performed on any violation in the media bandwidth policy. A system log is created by default. You can configure the no-syslog keyword to disable the Cisco Unified Border Element (Cisco UBE) from generating system logs on DSCP policy violation.
Configure a high value for DSCP violations. If you configure a low value such as 5, action will be performed on the call after every five violations and system logs will be generated frequently.
vmwi
To enable DC voltage or FSK visual message-waiting indictator (VMWI) on a Cisco VG224 onboard analog FXS voice port, use the vmwi command in voice-port configuration mode. To reset VMWI to default, use the no form of this command.
Usage Guidelines
This command with the dc-voltage keyword enables the message-waiting lamp to flash on an analog phone that requires DC voltage to activate a visual indicator.
This command with the fsk keyword enables the message-waiting lamp to flash on an analog phone that requires an FSK message to activate a visual indicator.
DC Voltage VMWI is supported for the SCCP telephony control (STC) application only. For all other applications, such as MGCP, FSK will be used even if you configure the vmwi dc-voltage command on the voice gateway.
vofr
To enable Voice over Frame Relay (VoFR) on a specific data-link connection identifier (DLCI) and to configure specific subchannels on that DLCI, use the vofr command in frame relay DLCI configuration mode. To disable VoFR on a specific DLCI, use the no form of this command.
Switched Calls
vofr [ data cid ] [ call-control [cid] ]
no vofr [ data cid ] [ call-control [cid] ]
Switched Calls to Cisco MC3810 Multiservice Concentrators Running Cisco IOS Releases Release Before 12.0(7)XK and Release 12.1(2)T
vofr [cisco]
no vofr [cisco]
Cisco-Trunk Permanent Calls
vofr data cid call-control cid
no vofr data cid call-control cid
FRF.11 Trunk Calls
vofr [ data cid ] [ call-control cid ]
no vofr [ data cid ] [ call-control cid ]
Syntax Description
data
(Required for Cisco-trunk permanent calls. Optional for switched calls.) Selects a subchannel (CID) for data other than the default subchannel, which is 4.
cid
(Optional) Specifies the subchannel to be used for data. Range is from 4 to 255. The default is 4. If data is specified, enter a valid CID.
call-control
(Optional) Reserves a subchannel for call-control signaling.
cisco
(Optional) Cisco proprietary voice encapsulation for VoFR with data is carried on CID 4 and call-control on CID 5.
cid
(Optional) Specifies the subchannel to be used for call-control signaling. Valid range is from 4 to 255. The default is 5. If call-control is specified and a CID is not entered, the default CID is used.
Command History
Release
Modification
12.0(3)XG
This command was introduced on Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series routers and Cisco MC3810.
12.0(4)T
This command was integrated into Cisco IOS Release 12.0(4)T.
12.0(7)XK
The use of the cisco option was modified. Beginning in this release, use the cisco option only when configuring connections to Cisco MC3810 running Cisco IOS Releases before 12.0(7)XK and 12.1(2)T.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
Usage Guidelines
The table below lists the different options of the vofr command and which combination of options is used beginning in Cisco IOS Release 12.0(7)XK and Release 12.1(2)T.
Table 1 Combinations of the vofr Command Type of Call
Command Combination to Use
Switched call (user dialed or auto-ringdown) to other routers supporting VoFR
vofr [data cid] [call-control [cid]]1
Cisco-trunk permanent call (private-line) to other routers supporting VoFR
vofr data cid call-control cid
FRF.11 trunk call (private-line) to other routers supporting VoFR
vofr [data cid] [call-control cid]2
1 The recommended form of this command to use is vofr data 4 call-control 5 .2 For FRF.11 trunk calls, the call-control option is not required. It is required only if you mix FRF.11 trunk calls with other types of voice calls on the same PVC.Examples
The following example, beginning in global configuration mode, shows how to enable VoFR on serial interface 1/1, DLCI 100. The example configures CID 4 for data; no call-control CID is defined.
interface serial 1/1 frame-relay interface-dlci 100 vofrTo configure CID 4 for data and CID 5 for call-control (both defaults), enter the following command:
vofr call-controlTo configure CID 10 for data and CID 15 for call-control, enter the following command:
vofr data 10 call-control 15To configure CID 4 for data and CID 15 for call-control, enter the following command:
vofr call-control 15To configure CID 10 for data and CID 5 for call-control, enter the following command:
vofr data 10 call-controlTo configure CID 10 for data with no call-control, enter the following command:
vofr data 10voice
To enable voice resource pool services for resource pool management, use the voice command in service profile configuration mode. To disable voice services, use the no form of this command.
Examples
The following example shows that voice service is available and enables voice resource pool service using the voice command in service profile configuration mode:
Router(config)# resource-pool profile service voip Router(config-service-profile)# ? Service Profile Configuration Commands: default Set a command to its defaults exit Exit from resource-manager configuration mode help Description of the interactive help system modem Configure modem service parameters no Negate a command or set in its defaults voice Configure voice service parameters Router(config-service-profile)# voicevoicecap configure
To apply a voicecap on NextPort platforms, use the voicecap configurecommand in voice-port configuration mode. To remove a voicecap, use the no form of this command.
Usage Guidelines
The character value for the name argument must be identical to the value entered when you created the voicecap using the voicecap entry command.
voicecap entry
To create a voicecap, use the voicecap entry command in global configuration mode. To disable a voicecap, use the no form of this command.
Syntax Description
Usage Guidelines
This command configures firmware through voicecap strings. This command allows you to assign values to specific registers. Voicecaps are applied to specific voice ports at system startup.
The voicecap values can be entered in a DSP-recognizable format called raw format . They can also be entered in standard format , which allows you to use commonly accessible values, such as decibels.
Starting with Cisco IOS Release 12.4(4)XC, this command can be used to configure GSMAMR-NB codecs on Cisco AS5350XM and Cisco AS5400XM platforms. The register values for GSMAMR-NB are shown in the table below.
Table 2 GSMAMR-NB Register Values V-Reg #
Default
Description
Register Values and Additional Notes
0
0
Sets how Adaptive Multi-Rate (AMR) responds to an incoming codec mode request (CMR) that is not a member of the mode set.
0 = Drop the packet with the bad CMR. 1 = Ignore the CMR (do not change rates) but process the rest of the packet data normally. 2 = Change the rate to the highest rate in the mode set lower than the rate requested by the CMR.
1
0
Sets how AMR handles packets with a frame type (AMR rate) that is not a member of the mode set.
0 = Drop the packet with the bad frame-type. 1 = Attempt to decode the packet.
voice call capacity mir
To set the value for the minimum interval between reporting (MIR), use the voice call capacity mir command in global configuration mode. To turn off these attributes, use the no form of this command.
voice call { carrier | trunk-group | prefix } capacity mir seconds
no voice call { carrier | trunk-group | prefix } capacity mir
Usage Guidelines
Because the available circuit (AC) attribute of a destination is very dynamic, reporting of this attribute should be handled carefully. AC should be reported as frequently as possible so that the location server has better information about the resources. However, the location server should not be overwhelmed with too many updates.
All of the AC reporting, called the interesting point of AC, is performed when the specified event happens within the minimum interval between reporting (MIR) time since last reporting. This command sets the amount of time used for the interval to control the number of interesting points that are reported so not to overwhelm the location server with too many AC updates.
The seconds argument cannot be set higher than the time configured for the capacity update interval.
voice call capacity reporting
To turn on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity, use the voice call capacity reporting command in global configuration mode. To turn off the reporting, use the no form of this command.
voice call { carrier | trunk-group | prefix } capacity reporting { maxima | inflection }
no voice call { carrier | trunk-group | prefix } capacity reporting { maxima | inflection }
Usage Guidelines
The smoothed curve of the available circuits (AC) has maxima, minima, and inflection points. When the curve has reached these points, this represents a change in the call rate.
Maximum, minimum and inflection points are illustrated in the figure below.
Examples
The following example shows the reporting of the available capacity inflection point on the trunk group is turned on:
Router(config)# voice call trunk-group capacity reporting inflectionRelated Commands
Command
Description
voice call capacity mir
Sets the values for the minimum interval between reporting (MIR) and smoothing transition time for weight (STW).
voice call capacity timer interval
Sets the periodic interval for reporting capacity from carrier, trunk group, or prefix databases
voice call trigger hwm
Sets the value for percentage change, low water mark and high water mark in the available capacity in the trunk group or prefix databases.
voice call capacity stw
To set the value for smoothing transition time for weight (STW), use the voice call capacity stw command in global configuration mode. To turn off these attributes, use the no form of this command.
voice call { carrier | trunk-group | prefix } capacity stw seconds
no voice call { carrier | trunk-group | prefix } capacity stw
Usage Guidelines
Because the available circuit (AC) attribute of a destination is very dynamic, reporting of this attribute should be handled carefully. AC should be reported as frequently as possible so that the location server has better information about the resources. However, the location server should not be overwhelmed with too many updates.
A smoothing algorithm is applied to the quantity of AC being reported. This algorithm eliminates reporting of noise. The degree of smoothing can be configured with the voice call capacity stw command. This command sets the smoothing transition time for weight, which is the time it takes for current smoothed value of AC to come half way between the current smoothed value and the current instantaneous value of AC. Lower stw values speed the smoothed value of AC as it approaches the instantaneous value of AC. When stw is set to 0, the smoothed value is always equal to the instantaneous value of AC.
voice call capacity timer interval
To set the periodic interval for reporting capacity from carrier, trunk group, or prefix databases, use the voice call capacity timer interval command in global configuration mode. To turn off the interval, use the no form of this command.
voice call { carrier | trunk-group | prefix } capacity timer interval seconds
no voice call { carrier | trunk-group | prefix } capacity timer interval seconds
Usage Guidelines
For the reporting interval, a periodic timer called the capacity update timer handles updates of available circuit (AC) information and can be configured using the voice call capacity timer interval command. For example, if AC has changed since the last reporting, the AC is again reported when the capacity update timer expires.
Examples
The following example sets the timer interval for the prefixes set at 15 seconds:
Router(config)# voice call prefix capacity timer interval 15Related Commands
Command
Description
voice call capacity mir
Sets the values for the MIR and STW.
voice call capacity reporting
Turns on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity.
voice call trigger hwm
Sets the value for percentage change, low water mark and high water mark in the available capacity in the trunk group or prefix databases.
voice call convert-discpi-to-prog
To convert a disconnect message with a progress indicator (PI) to a progress message, use the voice call convert-discpi-to-prog command in global configuration mode. To return to the default condition, use the no form of this command.
voice call convert-discpi-to-prog [ tunnel-IEs | always [tunnel-IEs] ]
no voice call convert-discpi-to-prog
Command History
Release
Modification
12.2(1)
This command was introduced.
12.3(6)
The tunnel-1Es keyword was added.
12.3(4)XQ
The always keyword with the tunnel-IEs keywordwere added.
12.3(8)T
The always keyword with the tunnel-IEs keywordwere added.
12.3(9)
The always keyword with the tunnel-1Es keywordwere added.
Usage Guidelines
The voice call convert-discpi-to-prog command turns an ISDN disconnect message into a progress message. If you use the tunnel-IEskeyword, the information elements are not dropped when the disconnect message is converted to a progress message.
Examples
The following example changes a disconnect with PI to a progress message containing information elements (IEs):
voice call convert-discpi-to-prog tunnel-IEsThe following example changes a disconnect with PI to a progress message in the preconnected and connected states:
voice call convert-discpi-to-prog alwaysvoice call csr data-points
To set the number of call success rate (CSR) data points, use the voice call csr data-points command in global configuration mode. To disable the setting of the CSR data points, use the no form of this command.
voice call { carrier | trunk-group | prefix } csr data-points value
no voice call { carrier | trunk-group | prefix } csr data-points value
voice call csr recording interval
To set the recording interval for call success rates (CSR), use the voice call csr recording interval command in global configuration mode. To disable the CSR recording interval, use the no form of this command.
voice call { carrier | trunk-group | prefix } csr recording interval minutes
no voice call { carrier | trunk-group | prefix } csr recording interval minutes
voice call csr reporting interval
To set the reporting interval for call success rate (CSR), use the voice call csr reporting interval command in global configuration mode. To disable the CSR recording interval, use the no form of this command.
voice call { carrier | trunk-group | prefix } csr reporting interval seconds
no voice call { carrier | trunk-group | prefix } csr reporting interval seconds
voice call debug
To debug a voice call, use the voice call debug command in global configuration mode. To disable the short-header setting and return tothe full-guid setting, use the no form of this command.
Command History
Release
Modification
12.2(11)T
The new debug header was added to the following platforms: Cisco 2600 series, Cisco 3620, Cisco 3640, Cisco 3660 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, Cisco AS5850, and Cisco MC3810.
12.2(15)T
The header-only keyword was replaced by the short-header keyword.
Usage Guidelines
Despite its nontraditional syntax (trailing rather than preceding "debug"), this is a normal debug command.
You can control the contents of the standardized header. Display options for the header are as follows:
The format of the GUID headers is as follows: //CallEntryID/GUID/Module-Dependent-List/Function-name:.
The format of the short header is as follows: //CallEntryID/Function-name:.
When the voice call debug short-header command is entered, the header displays with no GUID or module-specific parameters. When the no voice call debug short-header command is entered, the header, the 6-byte GUID, and module-dependent parameter output displays. The default option is displaying the 6-byte GUID trace.
NoteUsing the no form of this command does not turn off debugging.
Examples
The following is sample output when the full-guid keyword is specified:
Router# voice call debug full-guid ! 00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/vtsp_insert_cdb: 00:05:12: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/CCAPI/cc_incr_if_call_volume: 00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/vtsp_open_voice_and_set_params: 00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/vtsp_modem_proto_from_cdb: 00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/set_playout_cdb: 00:05:12: //1/0E2C8A90-BC00-11D5-8002-DACCFDCEF87D/VTSP:(0:D):0:0:4385/vtsp_dsp_echo_canceller_control:
NoteThe "//-1/" output indicates that CallEntryID for the CCAPI module is not available.
The table below describes significant fields shown in the display.
Table 3 voice call debug full-guid Field Descriptions Field
Description
VTSP:(0:D):0:0:4385
VTSP module, port name, channel number, DSP slot, and DSP channel number.
vtsp_insert_cdb
Function name.
CCAPI
CCAPI module.
The following is sample output when the short-header keyword is specified:
Router(config)# voice call debug short-header ! 00:05:12: //1/vtsp_insert_cdb: 00:05:12: //-1/cc_incr_if_call_volume: 00:05:12: //1/vtsp_open_voice_and_set_params: 00:05:12: //1/vtsp_modem_proto_from_cdb: 00:05:12: //1/set_playout_cdb: 00:05:12: //1/vtsp_dsp_echo_canceller_control:
NoteThe "//-1/" output indicates that CallEntryID for CCAPI is not available.
Related Commands
Command
Description
debug rtsp api
Displays debug output for the RTSP client API.
debug rtsp client session
Displays debug output for the RTSP client data.
debug rtsp error
Displays error message for RTSP data.
debug rtsp pmh
Displays debug messages for the PMH.
debug rtsp socket
Displays debug output for the RTSP client socket data.
debug voip ccapi error
Traces error logs in the CCAPI.
debug voip ccapi inout
Traces the execution path through the CCAPI.
debug voip ivr all
Displays all IVR messages.
debug voip ivr applib
Displays IVR API libraries being processed.
debug voip ivr callsetup
Displays IVR call setup being processed.
debug voip ivr digitcollect
Displays IVR digits collected during the call.
debug voip ivr dynamic
Displays IVR dynamic prompt play debug.
debug voip ivr error
Displays IVR errors.
debug voip ivr script
Displays IVR script debug.
debug voip ivr settlement
Displays IVR settlement activities.
debug voip ivr states
Displays IVR states.
debug voip ivr tclcommands
Displays the TCL commands used in the script.
debug voip rawmsg
Displays the raw VoIP message.
debug vtsp all
Enables debug vtsp session, debug vtsp error, and debug vtsp dsp.
debug vtsp dsp
Displays messages from the DSP.
debug vtsp error
Displays processing errors in the VTSP.
debug vtsp event
Displays the state of the gateway and the call events.
debug vtsp port
Limits VTSP debug output to a specific voice port.
debug vtsp rtp
Displays the voice telephony RTP packet debugging.
debug vtsp send-nse
Triggers the VTSP software module to send a triple redundant NSE.
debug vtsp session
Traces how the router interacts with the DSP.
debug vtsp stats
Debugs periodic statistical information sent and received from the DSP
debug vtsp vofr subframe
Displays the first 10 bytes of selected VoFR subframes for the interface.
debug vtsp tone
Displays the types of tones generated by the VoIP gateway.
voice call disc-pi-off
To enable the gateway to treat a disconnect message with progress indicator (PI) like a standard disconnect without a PI, use the voice call disc-pi-offcommand in global configuration mode. To reset to the default, use the no form of this command.
Command Default
Gateway disconnects incoming call leg when it receives a disconnect message with PI.
Usage Guidelines
Use this command if the gateway is connected to a switch that sends a release immediately after it receives a Disconnect with PI. To properly handle the call, the switch should open a backward voice path and keep the call active. Otherwise the rotary dial peer feature does not work because the incoming call leg is disconnected. Using this command enables the gateway to handle a disconnect with PI like a regular disconnect message so that you can use the rotary dial peer feature.
voice call rate monitor
To enable voice call rate monitoring, use the voice call rate monitor command in voice service configuration mode. To disable voice call monitoring, use the no form of this command.
Usage Guidelines
You can use the voice call rate monitor command to enable the call monitoring functionality for a duration of 60 seconds.
voice call send-alert
To enable the terminating gateway to send an alert message instead of a progress message after it receives a call setup message, use the voice call send-alert command inglobal configuration mode. To reset to the default, use the no form of this command.
Command Default
The terminating gateway sends a progress message after it receives a call Setup message.
Usage Guidelines
In Cisco IOS Release 12.1(3)XI and later, the terminating gateway sends a Progress message with a progress indicator (PI) after it receives a Setup message. Previously, the gateway responded with an Alert message after receiving a call. In some cases, if the terminating switch does not forward the progress message to the originating gateway, the originating gateway does not cut-through the voice path until a Connect is received and the caller does not hear a ringback tone. In these cases, you can use the voice call send-alert command to make the gateway backward compatible with releases earlier than Cisco IOS Release 12.1(3)XI. If you configure the voice call send-alert command, the terminating gateway sends an Alert message after it receives a Setup message from the originating gateway.
To complete calls from a PRI to an FXS interface, configure the voice call send-alert command on the FXS device.
voice call trap deviation
To configure the percentage deviation for voice call trap parameters, use the voice call trap deviation command in global configuration mode. To disable the configured percentage deviation, use the no form of this command.
Command Default
This command is enabled by default, and the deviation for trapping calls is set to 49 percent.
Usage Guidelines
Prior to Release 15.0(1)M, if a non-default percent value was configured, it could be disabled by entering the no voice call trap deviation percentcommand, even if the percent value was not the configured value. For example, if the voice call trap deviation 30 command was configured, the no voice call trap deviation 40command disabled the initial command.
Beginning in Release 15.0(1)M, the percent value in the no form of the command must match the configured non-default value. For example, if the voice call trap deviation 30 command is configured, the only way to disable it is to enter the no voice call trap deviation 30command. If the no voice call trap deviation 40 command is entered, the command-line interface displays this message: "Please enter correct deviation."
voice call trigger hwm
To set the value for high water mark in the available capacity in the trunk group or prefix databases, use the voice call trigger hwm command in global configuration mode. To disable the trigger point, use the no form of this command.
voice call { carrier | trunk-group | prefix } trigger hwm percent
no voice call { carrier | trunk-group | prefix } trigger hwm percent
Usage Guidelines
Available circuits are reported when the value of AC goes above a threshold, called the high water mark. This can be configured with the voice call trigger hwm command. When the hwm option is selected and the value is set to 100, no update is sent due to high water mark.
Examples
The following example sets the trigger for available capacity on trunk groups to send at a high water mark of 75%:
Router(config)# voice call trunk-group trigger hwm 75Related Commands
Command
Description
voice call capacity mir
Sets the values for the minimum interval between reporting (MIR) and smoothing transition time for weight (STW).
voice call capacity reporting
Turns on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity.
voice call capacity timer interval
Sets the periodic interval for reporting capacity from carrier, trunk group, or prefix databases
voice call trigger lwm
Sets the value for low water mark in the available capacity for carrier, trunk group, or prefix databases
voice call trigger percent-change
Sets the value for percentage change in the available capacity for carrier, trunk group, or prefix databases
voice call trigger lwm
To set the value for low water mark in the available capacity in the trunk group or prefix databases, use the voice call trigger lwm command in global configuration mode. To disable the trigger point, use the no form of this command.
voice call { carrier | trunk-group | prefix } trigger lwm percent
no voice call { carrier | trunk-group | prefix } trigger lwm percent
Usage Guidelines
Available circuits are reported when the value of AC falls below a threshold, called the low water mark. When the lwm option is selected and the value is set to 0, no update is sent due to low water mark.
Examples
The following example sets the trigger for available capacity for E.164 prefixes to send at a low water mark of 25%:
Router(config)# voice call prefix trigger lwm 25Related Commands
Command
Description
voice call capacity mir
Sets the values for the minimum interval between reporting (MIR) and smoothing transition time for weight (STW).
voice call capacity reporting
Turns on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity.
voice call capacity timer interval
Sets the periodic interval for reporting capacity from carrier, trunk group, or prefix databases.
voice call trigger hwm
Sets the value for high water mark in the available capacity for carrier, trunk group, or prefix databases.
voice call trigger percent-change
Sets the value for percentage change in the available capacity for carrier, trunk group, or prefix databases.
voice call trigger percent-change
To set the value for percentage change, low water mark and high water mark in the available capacity in the trunk group or prefix databases, use the voice call trigger command in global configuration mode. To disable the trigger point, use the no form of this command.
voice call { carrier | trunk-group | prefix } trigger percent-change percent
no voice call { carrier | trunk-group | prefix } trigger percent-change percent
Syntax Description
carrier
Carrier code address family
trunk-group
Trunk group address family
prefix
E.164 prefix
percent
If percent-change is selected, value can be 0 to 100 percent with a default of 30. If set to 0, this trigger will be turned off.
If lwm is selected, value can be 0 to 30 percent with a default of 10. If set to 0, this trigger will be turned off.
If hwm is select, value can be 50 to 100 percent with a default of 80. If set to 100, this trigger will be turned off.
Usage Guidelines
Available circuits are reported when the absolute percent change is above a threshold. When the percent-change option is selected and the value is set to 0, no update for percent change is sent
Examples
The following example sets the trigger for available capacity on the carrier codes to send at a percentage change of 15%:
Router(config)# voice call carrier trigger percent-change 15Related Commands
Command
Description
voice call capacity mir
Sets the values for the minimum interval between reporting (MIR) and smoothing transition time for weight (STW).
voice call capacity reporting
Turns on the reporting of maxima (first derivative) or inflection (second derivative) points in available capacity.
voice call capacity timer interval
Sets the periodic interval for reporting capacity from carrier, trunk group, or prefix databases
voice call trigger hwm
Sets the value for high water mark in the available capacity for carrier, trunk group, or prefix databases
voice call trigger lwm
Sets the value for low water mark in the available capacity for carrier, trunk group, or prefix databases
voice-card
To enter voice-card configuration mode and configure a voice card, use the voice-card command in global configuration mode. There is no no form of this command.
Syntax Description
Command History
Release
Modification
12.0(5)XK
The command was introduced on the Cisco 2600 series and Cisco 3600 series.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T.
12.0(7)XK
This command was implemented on the Cisco MC3810.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
12.2(2)XB
Values for the slot argument were updated to include AIMs.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T.
12.2(13)T
This command was supported in Cisco IOS Release 12.2(13)T and implemented on the Cisco 1700 series, Cisco 2600XM, Cisco 3700 series, Cisco 7200 series, Cisco 7500 series, Cisco ICS7750, Cisco MC3810, and Cisco VG200.
12.2(15)T
This command was integrated into Cisco IOS Release 12.2(15)T.
Usage Guidelines
Voice-card configuration mode is used for commands that configure the use of digital signal processing (DSP) resources, such as codec complexity and DSPs. DSP resources can be found in digital T1/E1 packet voice trunk network modules on Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series.
Codec complexity is configured in voice-card configuration mode and has the following platform-specific usage guidelines:
- On Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, and Cisco 3745, the slot argument corresponds to the physical chassis slot of the network module that has DSP resources to be configured.
DSP resource sharing is also configured in voice-card configuration mode. On the Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, and Cisco 3745 under specific circumstances, configuration of the dspfarm command enters DSP resources on a network module or AIM into a DSP resource pool. Those DSP resources are then available to process voice traffic on a different network module or voice/WAN interface card (VWIC). See the dspfarm (voice-card) command reference for more information about DSP resource sharing.
NoteWhen running high-complexity images, the system can only process up to 16 voice channels. Those 16 time slots need to be within a contiguous range (timeslot maximum (TSmax) minus timeslot minimum (TSmin) is less than or equal to 16, where TSmax and TSmin are the maximum DS0 and minimum DS0 configured for voice).
This command does not have a no form.
Examples
The following example enters voice-card configuration mode to configure resources on the network module in slot 1:
voice-card 1The following example shows how to enter voice-card configuration mode and load high-complexity DSP firmware on voice-card 0. The dspfarm command enters the DSP resources on the AIM specified in the voice-card command into the DSP resource pool.
voice-card 0 codec complexity high dspfarmvoice cause-code
To set the internal Q850 cause code mapping for voice and to enter voice cause configuration mode, use the voice cause-codecommand in global configuration mode. To disable the internal Q850 cause code mapping for voice, use the no form of this command.
voice class aaa
To enable dial-peer-based VoIP AAA configurations, use the voice class aaa command in global configuration mode. To disable dial-peer-based VoIP AAA configurations, use the no form of this command.
Usage Guidelines
The voice class aaa configuration command sets up a voice service class that allows you to perform dial-peer-based AAA configurations.
The command activates voice class AAA configuration mode. Commands that are configured in voice class AAA configuration mode are listed in the "Related Commands" section.
Examples
The following example shows AAA configurations in voice class AAA configuration mode. The number assigned to the tag is 1.
voice class aaa 1 authentication method dp authorization method dp accounting method dp in-bound accounting template temp-dpThe following example shows accounting configurations in voice class AAA configuration mode:
voice class aaa 2 accounting method dp-out out-bound accounting template temp-dp out-boundRelated Commands
Command
Description
accounting suppress
Disables accounting that is automatically generated by the service provider module for a specific dial peer.
authentication method
Specifies an authentication method for calls coming into the defined dial peer.
authorization method
Specifies an authorization method for calls coming into the defined dial peer.
method
Specifies an accounting method for calls coming into the defined dial peer.
voice-class aaa
Applies properties defined in the voice class to a specific dial peer.
voice class busyout
To create a voice class for local voice busyout functions, use the voice class busyout command inglobal configuration mode. To delete the voice class, use the no form of this command.
Usage Guidelines
You can apply a busyout voice class to multiple voice ports. You can assign only one busyout voice class to a voice port. If a second busyout voice class is assigned to a voice port, the second voice class replaces the one previously assigned.
If you assign a busyout voice class to a voice port, you may not assign separate busyout commands directly to the voice port, such as busyout monitor serial, busyout monitor ethernet, or busyout monitor probe.
Examples
The following example configures busyout voice class 20, in which the connections to two remote interfaces are monitored by a response time reporter (RTR) probe with a G.711ulaw profile, and voice ports are busied out whenever both links have a packet loss exceeding 10 percent and a packet delay time exceeding 2 seconds:
voice class busyout 20 busyout monitor probe 171.165.202.128 g711u loss 10 delay 2000 busyout monitor probe 171.165.202.129 g711u loss 10 delay 2000The following example configures busyout voice class 30, in which voice ports are busied out when serial ports 0/0, 1/0, 2/0, and 3/0 go out of service.
voice class busyout 30 busyout monitor serial 0/0 busyout monitor serial 1/0 busyout monitor serial 2/0 busyout monitor serial 3/0Related Commands
Command
Description
busyout monitor ethernet
Configures a voice port to monitor a local Ethernet interface for events that would trigger a voice-port busyout.
busyout monitor probe
Configures a voice port to enter the busyout state if an RTR probe signal returned from a remote, IP-addressable interface crosses a specified delay or loss threshold.
busyout monitor serial
Configures a voice port to monitor a serial interface for events that would trigger a voice-port busyout.
show voice busyout
Displays information about the voice busyout state.
voice class called number
To define a voice class called number or range of numbers, use the voice class called numbercommand in global configuration mode. To remove a voice class called number, use the no form of this command.
Usage Guidelines
Use this command to define one or more static voice class called numbers for inbound and outbound POTS dial peers or a dynamic voice class called number pool. The indexes for a voice class called number are defined with the index (voice class) command.
NoteEnter the voice class called number command in global configuration mode without hyphens. Enter the voice-class called-number command in dial-peer configuration mode with hyphens.
Examples
The following example shows configuration for an outbound voice class called number:
voice class called number outbound 30 index 1 5550100 index 2 5550101 index 3 5550102 index 4 5550103The following example shows configuration for a voice class called number pool:
voice class called number pool 1 index 1 5550100 - 5550199voice class cause-code
To configure cause code list parameters for a voice class and to enter cause code configuration mode, use the voice class cause-codecommand in global configuration mode. To disable the cause code list parameters configuration for a voice class, use the no form of this command.
voice class codec
To enter voice-class configuration mode and assign an identification tag number for a codec voice class, use the voice class codec command in global configuration mode. To delete a codec voice class, use the no form of this command.
Usage Guidelines
This command only creates the voice class for codec selection preference and assigns an identification tag. Use the codec preference command to specify the parameters of the voice class, and use the voice-class codec dial-peer command to apply the voice class to a VoIP dial peer.
NoteThe voice class codeccommand in global configuration mode is entered without a hyphen. The voice-class codeccommand in dial-peer configuration mode is entered with a hyphen.
Examples
The following example shows how to enter voice-class configuration mode and assign a voice class tag number starting from global configuration mode:
voice class codec 10After you enter voice-class configuration mode for codecs, use the codec preference command to specify the parameters of the voice class.
The following example creates preference list 99, which can be applied to any dial peer:
voice class codec 99 codec preference 1 g711alaw codec preference 2 g711ulaw bytes 80 codec preference 3 g723ar53 codec preference 4 g723ar63 bytes 144 codec preference 5 g723r53 codec preference 6 g723r63 bytes 120 codec preference 7 g726r16 codec preference 8 g726r24 codec preference 9 g726r32 bytes 80 codec preference 10 g728 codec preference 11 g729br8 codec preference 12 g729r8 bytes 50Related Commands
Command
Description
codec preference
Specifies a list of preferred codecs to use on a dial peer.
test voice port detector
Defines the order of preference in which network dial peers select codecs.
voice-class codec (dial peer)
Assigns a previously configured codec selection preference list to a dial peer.
voice class custom-cptone
To create a voice class for defining custom call-progress tones to be detected, use the voice class custom-cptone command inglobal configuration mode. To delete the voice class, use the no form of this command.
Usage Guidelines
After you create a voice class, you need to define custom call-progress tones for this voice class using the dualtone command.
voice class dscp-profile
To configure the differentiated services code point (DSCP) profile, use the voice class dscp-profile command in global configuration mode. To disable the configuration, use the no form of this command.
Usage Guidelines
You can use the voice class dscp-profile command to configure the DSCP profile and then configure DSCP policing and enter voice class configuration mode.
voice class dualtone
To create a voice class for Foreign Exchange Office (FXO) supervisory disconnect tone detection parameters, use the voice class dualtonecommand in global configuration mode. To delete the voice class, use the no form of this command.
Usage Guidelines
Use this command first to create the voice class. Then use the supervisory disconnect dualtone voice-classcommand to assign the voice class to a voice port.
A voice class can define any number of tones to be detected. You need to define a matching tone for each supervisory disconnect tone expected from a PBX or from the public switched telephone network (PSTN).
Examples
The following example configures voice class dualtone 70, which defines one tone with two frequency components, and does not configure a cadence list:
voice class dualtone 100 freq-pair 1 350 440 freq-max-deviation 10 freq-max-power 6 freq-min-power 25 freq-power-twist 15 freq-max-delay 16 cadence-min-on-time 50 cadence-max-off-time 400 cadence-variation 8 exitThe following example configures voice class dualtone 100, which defines one tone with two frequency components, and configures a cadence list:
voice class dualtone 100 freq-pair 1 350 440 freq-pair 2 480 850 freq-max-deviation 10 freq-max-power 6 freq-min-power 25 freq-power-twist 15 freq-max-delay 16 cadence-min-on-time 50 cadence-max-off-time 400 cadence-list 1 100 100 300 300 cadence-variation 8 exitThe following example configures voice class dualtone 90, which defines three tones, each with two frequency components, and configures two cadence lists:
voice class dualtone 90 freq-pair 1 350 440 freq-pair 2 480 850 freq-pair 3 1000 1250 freq-max-deviation 10 freq-max-power 6 freq-min-power 25 freq-power-twist 15 freq-max-delay 16 cadence-min-on-time 50 cadence-max-off-time 500 cadence-list 1 100 100 300 300 100 200 cadence-list 2 100 200 100 400 cadence-variation 8 exitvoice class dualtone-detect-params
To create a voice class for defining a set of tolerance limits for the frequency, power, and cadence parameters of the tones to be detected, use the voice class dualtone-detect-params command inglobal configuration mode. To delete the voice class, use the no form of this command.
Usage Guidelines
Use this command to create a voice class in which you can define maximum and minimum call-progress tone tolerance parameters that you can apply to any voice port. These parameters further define the call-progress tones defined by the voice class custom-cptone command. Use the supervisory dualtone-detect-paramscommand to apply these tolerance parameters to a voice port.
Examples
The following example creates voice class 70, in which you can specify modified boundaries and limits for call-progress tone detection.
voice class dualtone-detect-params 70 freq-max-deviation 25 freq-max-power -5 freq-min-power -20 freq-power-twist 10 freq-max-delay 50 cadence-variation 80 exitvoice class e164-pattern-map
To create an E.164 pattern map that specifies multiple destination E.164 patterns in a dial peer, use the voice class e164-pattern map command in global configuration mode. To remove an E.164 pattern map from a dial peer, use the no form of this command.
voice class e164-pattern-map load
To load a destination E.164 pattern map that is specified by a text file on a dial peer, use the voice class e164-pattern-map load command in privileged EXEC mode.
Usage Guidelines
After creating an E.164 pattern map, you can add destination E.164 pattern entries to the E.164 pattern map and store all the information on the voice gateway or create the E.164 pattern entries in a text file and store the file on the internally or externally supported file system.
voice class h323
To create an H.323 voice class that is independent of a dial peer and can be used on multiple dial peers, use the voice class h323 command in global configuration mode. To remove the voice class, use the no form of this command.
Usage Guidelines
The voice class h323 command in global configuration mode does not include a hyphen. The voice-class h323 command in dial-peer configuration mode includes a hyphen.
Examples
The following example demonstrates how a voice class is created and applied to an individual dial peer. Voice class 4 contains a command to disable the capability to detect Cisco CallManager systems in the network (this command is used by Cisco CallManager Express 3.1 and later versions). The example then uses the voice-class h323command to apply voice class 4 to dial peer 36.
Router(config)# voice class h323 4 Router(config-class)# no telephony-service ccm-compatible Router(config-class)# exit Router(config)# dial-peer voice 36 voip Router(config-dial-peer)# destination-pattern 555.... Router(config-dial-peer)# session target ipv4:10.5.6.7 Router(config-dial-peer)# voice-class h323 4voice class media
To configure the media control parameters for voice, use the voice class mediacommand in global configuration mode. To disable the media control parameters for voice, use the no form of this command.
voice class permanent
To create a voice class for a Cisco trunk or FRF.11 trunk, use the voice class permanent command in global configuration mode. To delete the voice class, use the no form of this command.
Usage Guidelines
The voice class permanent command can be used for Voice over Frame Relay (VoFR), Voice over ATM (VoATM), and Voice over IP (VoIP) trunks.
The voice class permanentcommand in global configuration mode is entered without a hyphen. The voice-class permanentcommand in dial-peer and voice-port configuration modes is entered with a hyphen.
Examples
The following example shows how to create a permanent voice class starting from global configuration mode:
voice class permanent 10 signal keepalive 3 exitRelated Commands
Command
Description
signal keepalive
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
signal pattern
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
signal timing idle suppress-voice
Configures the signal timing parameter for the idle state of a call.
signal timing oos
Configures the signal timing parameter for the OOS state of a call.
signal-type
Sets the signaling type for a network dial peer.
voice-class permanent
Assigns a previously configured voice class for a Cisco trunk or FRF.11 trunk to a network dial peer.
voice class resource-group
To enter voice-class configuration mode and assign an identification tag number for a resource group, use the voice class resource-groupcommand in global configuration mode. To delete a resource group, use the no form of this command.
Usage Guidelines
Use the voice class resource-group command to configure parameters along with the threshold values to be monitored for resource groups. When you use the voice class resource-group command, the router enters voice-class configuration mode. You can then group the resources to be monitored and configure parameters such as .
Examples
The following example shows how to enter voice-class configuration mode and assign identification tag number 5 for a resource group:
Router> enable Router# configure terminal Router(config)# voice class resource-group 5Related Commands
Command
Description
debug rai
Enables debugging for Resource Allocation Indication (RAI).
periodic-report interval
Configures periodic reporting parameters for gateway resource entities.
rai target
Configures the SIP RAI mechanism.
resource (voice)
Configures parameters for monitoring resources.
show voice class resource-group
Displays the resource group configuration information for a specific resource group or all resource groups.
voice class sip-copylist
To configure a list of entities to be sent to the peer call leg, use the voice class sip-copylist command in global configuration mode. To disable the configuration, use the no form of this command.
Usage Guidelines
Use the voice class sip-copylist command to configure Cisco Unified Border Element (UBE) to pass an unsupported parameter present in a mandatory header from one call leg to another of Cisco UBE. You can copy the inbound message headers into variables and pass the headers to the outbound call leg.
voice class sip-profiles
To configure Session Initiation Protocol (SIP) profiles for a voice class, use the voice class sip-profilescommand in global configuration mode. To disable the SIP profiles for a voice class, use the no form of this command.
voice class tone-signal
To enter voice-class configuration mode and create a tone-signal voice class, use the voice class tone-signal command in global configuration mode. To delete a tone-signal voice class, use the no form of this command.
Usage Guidelines
Use the voice class tone-signal command to define wakeup, frequency selection, and guard tones to be played out before and during the voice packets for a specific voice port. Use the inject guard-tone, inject pause, and inject tone commands to define the tone signaling in this class. You can configure up to ten tones in a tone-signal voice class.
To avoid voice loss at the receiving end of an LMR system, the maximum of the sum of the durations of the injected tones and pauses in the voice class should not exceed 1500 milliseconds. You must also use the timing delay-voice tdm command to configure a delay for the voice packet equal to the sum of the durations of all the injected tones and pauses.
Note that the hyphenation in this command differs from the hyphenation used in a similar command, voice-class tone-signal, which is used in voice-port configuration mode.
Examples
The following example shows how to create a tone-signal voice class starting from global configuration mode:
voice class tone-signal mytones inject tone 1 1950 3 150 inject tone 2 2000 0 60 inject pause 3 60 inject tone 4 2175 3 150 inject tone 5 1000 0 50Related Commands
Command
Description
inject guard-tone
Plays out a guard tone with the voice packet.
inject pause
Specifies a pause between injected tones.
inject tone
Specifies a wakeup or frequency selection tone to be played out before the voice packet.
timing delay-voice tdm
Specifies the delay before a voice packet is played out.
voice-class tone-signal
Assigns a previously configured tone-signal voice class to a voice port.
voice class uri
To create or modify a voice class for matching dial peers to a Session Initiation Protocol (SIP) or telephone (TEL) uniform resource identifier (URI), use the voice class uricommand in global configuration mode. To remove the voice class, use the no form of this command.
Usage Guidelines
- This command takes you to voice URI class configuration mode, where you configure the match characteristics for a URI. The commands that you enter in this mode define the set of rules by which the URI in a call is matched to a dial peer.
- To reference this voice class for incoming calls, use the incoming uri command in the inbound dial peer. To reference this voice class for outgoing calls, use the destination uri command in the outbound dial peer.
- Using the no voice class uri command removes the voice class from any dial peer where it is configured with the destination uri or incoming uri commands.
Examples
The following example defines a voice class for SIP URIs:
voice class uri r100 sip user-id abc123 host server1 phone context 408The following example defines a voice class for TEL URIs:
voice class uri r101 tel phone number ^408 phone context 408Related Commands
Command
Description
debug voice uri
Displays debugging messages related to URI voice classes.
destination uri
Specifies the voice class used to match the dial peer to the destination URI for an outgoing call.
host
Matches a call based on the host field in a SIP URI.
incoming uri
Specifies the voice class used to match a VoIP dial peer to the URI of an incoming call.
pattern
Matches a call based on the entire SIP or TEL URI.
phone context
Filters out URIs that do not contain a phone-context field that matches the configured pattern.
phone number
Matches a call based on the phone number field in a TEL URI.
show dialplan incall uri
Displays which dial peer is matched for a specific URI in an incoming call.
show dialplan uri
Displays which outbound dial peer is matched for a specific destination URI.
user-id
Matches a call based on the user-id field in the SIP URI.
voice class uri sip preference
To set the preference for selecting a voice class for Session Initiation Protocol (SIP) uniform resource identifiers (URIs), use the voice class uri sip preferencecommand in global configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
- Use this command to resolve ties when more than one voice class is matched for a SIP URI. The default is to match on the host field of the URI.
- This command applies globally to all URI voice classes for SIP.
Examples
The following example defines the preference as the user-id for a SIP voice class:
voice class uri sip preference user-idRelated Commands
Command
Description
debug voice uri
Displays debugging messages related to URI voice classes.
destination uri
Specifies the voice class used to match the dial peer to the destination URI for an outgoing call.
host
Matches a call based on the host field in a SIP URI.
incoming uri
Specifies the voice class used to match a VoIP dial peer to the URI of an incoming call.
user-id
Matches a call based on the user-id field in the SIP URI.
show dialplan incall uri
Displays which dial peer is matched for a specific URI in an incoming call.
show dialplan uri
Displays which outbound dial peer is matched for a specific destination URI.
voice class uri
Creates or modifies a voice class for matching dial peers to a SIP or TEL URI.
voice-class aaa (dial peer)
To apply properties defined in the voice class to a dial peer, use the voice-class aaacommand in dial-peer configuration mode. This command does not have a no form.
Usage Guidelines
Properties that are configured in voice class AAA configuration mode can be applied to a dial peer by using this command.
Examples
The following example shows redirecting AAA requests using Digital Number Identification Service (DNIS). You define a voice class to specify the AAA methods and then use this command.
voice class aaa 1 authentication method kz authorization method kz accounting method kz ! dial-peer voice 100 voip incoming called-number 50.. session target ipv4:1.5.31.201 voice-class aaa 1voice-class called-number (dial peer)
To assign a previously defined voice class called number to an inbound or outbound POTS dial peer, use the voice-class called-numbercommand in dial peer configuration mode. To remove a voice class called number from the dial peer, use the no form of this command.
Usage Guidelines
Use this command to assign a previously defined voice class called number to a dial peer for a static H.320 secondary call dial plan. Use the inbound keyword for inbound POTS dial peers, and the outbound keyword for outbound POTS dial peers.
NoteThe voice class called number command in global configuration mode is entered without hyphens. The voice-class called-number command in dial peer configuration mode is entered with hyphens.
voice-class called-number-pool
To assign a previously defined voice class called number pool to a voice port, use the voice-class called-number-pool command in voice class configuration mode. To remove a voice class called number pool from the voice port, use the no form of this command.
Usage Guidelines
Use this command to assign a voice class called number pool to a voice port for a dynamic H.320 secondary call dial plan.
voice-class codec (dial peer)
To assign a previously configured codec selection preference list (codec voice class) to a VoIP dial peer, enter the voice-class codec command in dial-peer configuration mode. To remove the codec preference assignment from the dial peer, use the no form of this command.
Syntax Description
Command History
Release
Modification
12.0(2)XH
This command was introduced in Cisco IOS Release 12.0(2)XH and implemented on the Cisco AS5300 series routers.
12.0(7)T
This command was integrated into Cisco IOS Release 12.0(7)T and implemented on the Cisco 2600 series and the Cisco 3600 series.
12.0(7)XK
This command was integrated into Cisco IOS Release 12.0(7)XK and implemented on the Cisco MC3810.
15.1(2)T
This command was modified. The offer-all keyword was added.
Cisco IOS XE Release 2.5
This command was integrated into Cisco IOS XE Release 2.5.
Usage Guidelines
You can assign one voice class to each VoIP dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
NoteThe voice-class codec command in dial-peer configuration mode is entered with a hyphen. The voice class codec command in global configuration mode is entered without a hyphen.
Examples
The following example shows how to assign a previously configured codec voice class to a dial peer:
Router# configure terminal Router(config)# dial-peer voice 100 voip Router(config-dial-peer)# voice-class codec 10 offer-allRelated Commands
Command
Description
show dial-peer voice
Displays the configuration for all dial peers configured on the router.
test voice port detector
Defines the order of preference in which network dial peers select codecs.
voice class codec
Enters voice-class configuration mode and assigns an identification tag number for a codec voice class.
voice-class h323 (dial peer)
To assign an H.323 voice class to a VoIP dial peer, use the voice-class h323 command in dial-peer configuration mode. To remove the voice class from the dial peer, use the no form of this command.
Usage Guidelines
The voice class that you assign to the dial peer must be configured using the voice class h323 in global configuration mode.
You can assign one voice class to each VoIP dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
The voice-class h323 command in dial-peer configuration mode includes a hyphen and in global configuration mode does not include a hyphen.
Examples
The following example demonstrates how a voice class is created and applied to an individual dial peer. Voice class 4 contains a command to disable the capability to detect Cisco CallManager systems in the network (this command is used by Cisco CallManager Express 3.1 and later versions). The example then uses the voice-class h323command to apply voice class 4 to dial peer 36.
Router(config)# voice class h323 4 Router(config-class)# no telephony-service ccm-compatible Router(config-class)# exit Router(config)# dial-peer voice 36 voip Router(config-dial-peer)# destination-pattern 555.... Router(config-dial-peer)# session target ipv4:10.5.6.7 Router(config-dial-peer)# voice-class h323 4voice-class permanent (dial-peer)
To assign a previously configured voice class for a Cisco trunk or FRF.11 trunk to a network dial peer, use the voice-class permanent command in dial-peer configuration mode. To remove the voice-class assignment from the network dial peer, use the no form of this command.
Usage Guidelines
You can assign one voice class to any given network dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
You cannot assign a voice class to a plain old telephone service (POTS) dial peer.
The voice-class permanentcommand in dial-peer configuration mode is entered with a hyphen. The voice class permanentcommand in global configuration mode is entered without a hyphen.
Examples
The following example assigns a previously configured voice class to a Voice over Frame Relay (VoFR) network dial peer:
dial-peer voice 100 vofr voice-class permanent 10Related Commands
Command
Description
signal keepalive
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
signal pattern
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
signal timing idle suppress-voice
Configures the signal timing parameter for the idle state of a call.
signal timing oos
Configures the signal timing parameter for the OOS state of a call.
signal-type
Sets the signaling type for a network dial peer.
voice class permanent
Creates a voice class for a Cisco trunk or FRF.11 trunk.
voice-class permanent (voice-port)
To assign a previously configured voice class for a Cisco trunk or FRF.11 trunk to a voice port, use the voice-class permanent command in voice-port configuration mode. To remove the voice-class assignment from the voice port, use the no form of this command.
Usage Guidelines
You can assign one voice class to any given voice port. If you assign another voice class to a voice port, the last voice class assigned replaces the previous voice class.
The voice-class permanentcommand in voice-port configuration mode is entered with a hyphen. The voice class permanentcommand in global configuration mode is entered without a hyphen.
Examples
The following example assigns a previously configured voice class to voice port 1/1/0:
voice-port 1/1/0 voice-class permanent 10Related Commands
Command
Description
signal keepalive
Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks.
signal pattern
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks.
signal timing idle suppress-voice
Configures the signal timing parameter for the idle state of a call.
signal timing oos
Configures the signal timing parameter for the OOS state of a call.
signal-type
Sets the signaling type for a network dial peer.
voice class permanent
Creates a voice class for a Cisco trunk or FRF.11 trunk.
voice-class sip anat
To enable Alternative Network Address Types (ANAT) on a Session Initiation Protocol (SIP) trunk, use the voice-class sip anat command in SIP configuration or dial peer configuration mode. To disable ANAT on SIP trunks, use the no form of this command.
Usage Guidelines
Both the Cisco IOS SIP gateway and Cisco Unified Border Element are required to support the Session Description Protocol (SDP) ANAT semantics. The bind command allows the use of ANAT semantics in outbound SDP. SDP ANAT semantics are intended to address scenarios that involve different network address families (for example, different IPv4 versions). Media lines grouped using ANAT semantics provide alternative network addresses of different families for a single logical media stream. The entity creating a session description with an ANAT group must be ready to receive or send media over any of the grouped "m" lines.
By default, ANAT is enabled on SIP trunks. However, if the SIP gateway is configured in IPv4-only mode or IPv6-only mode, the gateway will not use ANAT semantics in its SDP offer.
The system keyword configures ANAT on all network dial peers, including the local dial peer. Using the voice-class sip anat command without the system keyword enables ANAT only for the local dial peer.
voice pcm capture
To allocate the number of Pulse Code Modulation (PCM) capture buffers, to set up or change the destination URL for captured data, to enable PCM capture on-demand, and to change the PCM capture trigger string by the user, use the voice pcm capture command in global configuration mode. To stop all logging and file operations, to disable data transport from the capture buffer, and to automatically set the number of buffers to 0, use the no form of this command.
voice pcm capture { buffer number | destination url | on-demand-trigger | user-trigger-string start-string stop-string stream bitmap duration call-duration }
no voice pcm capture { buffer number | destination url | on-demand-trigger | user-trigger-string }
Syntax Description
Usage Guidelines
If you want to change the number of an existing nonzero buffer, you must first reset it to 0 and then change it from 0 to the new number.
The destination url option sets up or changes the destination URL for captured data. To disable data transport from the capture buffer, use the no form of this command. If the buffer is allocated, captured data is sent to the current URL (if it was already configured) until the new URL is specified.
If a new URL differs from the current URL and logging is enabled, the current URL is closed and all further data is sent to the new URL. Entering a blank URL or prefixing the command with no disables data transport from the capture buffer, and (if capture is enabled) captured data is stored in the capture buffer until it reaches its capacity.
Once the buffer-queueing program is running, the transport process attempts to connect to a new or existing “capture destination” URL. A version message is written to the URL, and if the message is successfully received, any further messages placed into the message queue are written to that URL. If a new URL is entered using the voice pcm capture destination url command, the open URL is closed, and the system attempts to write to the new URL. If the new URL does not work, the transport process exits. The transport process is restarted when another URL is entered or the system is restarted.
Examples
The following example shows how to configure the number of PCM capture buffers:
Router> enable Router# configure terminal Router(config)# voice pcm capture buffer 200The following example shows how to configure the destination URL for storing captured data:
Router> enable Router# configure terminal Router(config)# voice pcm capture destination tftp://10.0.1.10/acphan/The following example shows how to configure user trigger PCM capture:
Router> enable Router# configure terminal Router(config)# voice pcm capture on-demand-triggerThe following example shows how to change the default user trigger PCM capture start and stop string, stream, and call duration:
Router> enable Router# configure terminal Router(config)# voice pcm capture #132 #543 stream ff duration 230voice-class sip asserted-id
To enable support for the dial-peer-based asserted ID header in incoming Session Initiation Protocol (SIP) requests or response messages, and to send asserted ID privacy information in outgoing SIP requests or response messages, use the voice-class sip asserted-id command in dial-peer configuration mode. To disable the support for the asserted ID header, use the no form of this command.
Syntax Description
pai
(Optional) Enables the P-Asserted-Identity (PAI) privacy header in incoming and outgoing SIP requests or response messages.
ppi
(Optional) Enables the P-Preferred-Identity (PPI) privacy header in incoming SIP requests and outgoing SIP requests or response messages.
system
(Optional) Uses global-level configuration settings to configure the dial peer.
Command Default
The privacy information is sent using the Remote-Party-ID (RPID) header or the FROM header.
Usage Guidelines
If you choose the pai keyword or the ppi keyword for incoming messages, the gateway builds the PAI or the PPI header, respectively, into the common SIP stack, thereby sending the call data using the PAI or the PPI header. For outgoing messages, the privacy information is sent on the PAI or PPI header. The pai keyword or the ppi keyword has priority over the Remote-Party-ID (RPID) header, and removes the RPID/FROM header from the outbound message, even if the router is configured to use the RPID header at the global level.
Examples
The following example shows how to enable support for the PPI header:
Router> enable Router# configure terminal Router(config)# dial peer voice 1 Router(conf-voi-serv)# voice-class sip asserted-id ppivoice-class sip associate registered-number
To associate the preloaded route and outbound proxy details to the registered number in the dail peer configuration mode, use the voice-class sip associate registered-numbercommand in dial peer configuration mode. To remove the association, use the no form of this command.
voice-class sip associate registered-number number [system]
no voice-class sip associate registered-number
Usage Guidelines
The voice-class sip associate registered-numbercommand takes precedence over the associate registered-numbercommand in voice service VOIP SIP configuration mode. However, if the voice-class sip associate registered-number command is used with the system keyword, the gateway uses the settings configured globally by the associate registered-numbercommand.
voice-class sip asymmetric payload
To configure dynamic Session Initiation Protocol (SIP) asymmetric payload support on a dial peer, use the voice-class sip asymmetric payloadcommand in dial peer configuration mode. To disable the configuration, use the no form of this command.
voice-class sip asymmetric payload { dtmf | dynamic-codecs | full | system }
no voice-class sip asymmetric payload
Syntax Description
dtmf
Provides asymmetric support only for dual-tone multi-frequency (DTMF) payloads.
dynamic-codecs
Provides asymmetric support only for dynamic codec payloads.
full
Provides asymmetric support both for DTMF and dynamic codec payloads.
system
(Optional) Specifies that the asymmetric payload uses the global value.
Usage Guidelines
For the Cisco UBE the SIP asymmetric payload-type is supported for audio/video codecs, DTMF, and NSE. Hence, dtmf and dynamic-codecs keywords are internally mapped to the full keyword to provide asymmetric payload-type support for audio/video codecs , DTMF, and NSE.
voice-class sip authenticate redirecting-number
To supersede global settings and enable a dial peer on a Cisco IOS voice gateway to authenticate and pass Session Initiation Protocol (SIP) credentials based on the redirecting number of forwarded calls, use the voice-class sip authenticate redirecting-number command in dial peer voice configuration mode. To supersede global settings and specify that a dial peer uses only the calling number of forwarded calls, use the no form of this command. To return a dial peer to the default setting so that the dial peer uses the global setting, use the default form of this command.
voice-class sip authenticate redirecting-number [system]
no voice-class sip authenticate redirecting-number
default voice-class sip authenticate redirecting-number
Command Default
The dial peer uses the global setting. If the global setting is not specifically configured, the dial peer uses only the calling number of a forwarded call for SIP credentials even when the redirecting number is available for that call.
Usage Guidelines
When an INVITE message sent out by the gateway is challenged, it must respond with the appropriate SIP credentials before the call is established. The default global behavior for the gateway is to authenticate and pass SIP credentials based on the calling number and all dial peers on a gateway default to the global setting. However, for forwarded calls, it is sometimes more appropriate to use the redirecting number and this can be specified at either the global or dial peer level (configuring behavior for a specific dial peer supersedes the global setting).
Use the voice-class sip authenticate redirecting-number command in dial peer voice configuration mode to supersede global settings and enable a dial peer to authenticate and pass SIP credentials based on the redirecting number when available. Use the no form of this command to supersede global settings and force a dial peer to authenticate and pass SIP credentials based only on the calling number of forwarded calls. Use the default form of this command to configure the dial peer to use the global setting.
The redirecting number is present only in the headers of forwarded calls. When the voice-class sip authenticate redirecting-number command is disabled or the redirecting number is not available, the dial peer passes SIP credentials that are based on the calling number of the forwarded call. This is also the behavior on dial peers that are configured to use the global setting and the global setting is disabled (default). To enable the global setting (which is used as the default setting for all dial peers on the gateway), use the authenticate redirecting-number command in voice service SIP configuration mode.
Examples
The following example shows how to enable dial peer 2 to authenticate and pass SIP credentials based on the redirecting number (if available) of a forwarded call when a SIP INVITE message is challenged:
Router> enable Router# configure terminal Router(config)# dial-peer voice 2 voip Router(config-dial-peer)# voice-class sip authenticate redirecting-numberThe following example shows how to force dial peer 2 to authenticate and pass only the calling number of a call even when the global setting is enabled and a redirecting number is available for a call:
Router> enable Router# configure terminal Router(config)# dial-peer voice 2 voip Router(config-dial-peer)# no voice-class sip authenticate redirecting-numberThe following two examples show different ways of setting dial peer 2 to the default setting so that it authenticates and passes either the redirecting or calling number of a call based on the global (system) setting for the gateway:
Router> enable Router# configure terminal Router(config)# dial-peer voice 2 voip Router(config-dial-peer)# default voice-class sip authenticate redirecting-number Router> enable Router# configure terminal Router(config)# dial-peer voice 2 voip Router(config-dial-peer)# voice-class sip authenticate redirecting-number systemvoice-class sip bind
To bind the source address of a specific interface for a dial-peer on a Session Initiation Protocol (SIP) trunk, use the voice-class sip bindcommand in dial peer voice configuration mode. To disable bind at the dial-peer level or restore the bind to the global level, use the no form of this command.
voice-class sip bind { control | media } source-interface interface-id [ ipv6-address ipv6-address ]
no voice-class sip bind { control | media }
Usage Guidelines
Use the voice-class sip bind command in dial peer voice configuration mode to bind the source address for signaling and media packets to the IP address of an interface on Cisco IOS voice gateway.
You can configure multiple IPv6 addresses for an interface and select one address using the ipv6-address keyword.
Examples
The following example shows how to configure SIP bind command:
Router(config)# dial-peer voice 101 voip Router(config-dial-peer)# session protocol sipv2 Router(config-dial-peer)# voice-class sip bind control source-interface GigabitEthernet0/0 ipv6-address 2001:0DB8:0:1::1 Router(config-dial-peer)# voice-class sip bind media source-interface GigabitEthernet0/0voice-class sip block
To configure an individual dial peer on a Cisco IOS voice gateway or Cisco Unified Border Element (Cisco UBE) to drop (not pass) specific incoming Session Initiation Protocol (SIP) provisional response messages, use the voice-class sip block command in dial peer voice configuration mode. To disable a configuration to drop incoming SIP provisional response messages on an individual dial peer, use the no form of this command.
voice-class sip block { 180 | 181 | 183 } [ sdp { absent | present } | system ]
no voice-class sip block { 180 | 181 | 183 }
Syntax Description
180
Specifies that incoming SIP 180 Ringing messages should be dropped (not passed to the other leg).
181
Specifies that incoming SIP 181 Call is Being Forwarded messages should be dropped (not passed to the other leg).
183
Specifies that incoming SIP 183 Session in Progress messages should be dropped (not passed to the other leg).
sdp
(Optional) Specifies that either the presence or absence of Session Description Protocol (SDP) information in the received response determines when the dropping of specified incoming SIP messages takes place.
absent
Configures the SDP option so that specified incoming SIP messages are dropped only if SDP is absent from the received provisional response.
present
Configures the SDP option so that specified incoming SIP messages are dropped only if SDP is present in the received provisional response.
system
Configures the dial peer to use global configuration settings for dropping incoming SIP provisional response messages.
Command Default
Defaults to the global configuration setting, which, when not specifically configured, means incoming SIP 180, 181, and 183 provisional responses are forwarded.
Command History
Release
Modification
12.4(22)YB
This command was introduced. Only SIP 180 and SIP 183 messages are supported on Cisco UBEs.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
15.0(1)XA
This command was modified. Support was added for SIP 181 messages on the Cisco IOS SIP gateway, SIP-SIP Cisco UBEs, and the SIP trunk of Cisco Unified Communications Manager Express (Cisco Unified CME).
15.1(1)T
This command was integrated into Cisco IOS Release 15.1(1)T.
Cisco IOS XE Release 3.1S
This command was integrated into Cisco IOS XE Release 3.1S.
Usage Guidelines
Use the voice-class sip block command in dial peer voice configuration mode to configure a specific dial peer on a Cisco IOS voice gateway or Cisco UBE to override global settings and drop specified SIP provisional response messages. Additionally, you can use the sdp keyword to further control when the specified SIP message is dropped based on either the absence or presence of SDP information.
You can also use the system keyword to configure a specific dial peer to use global configuration settings for dropping incoming SIP provisional response messages. To configure global settings on a Cisco IOS voice gateway or Cisco UBE, use the block command in voice service SIP configuration mode. To disable configurations for dropping specified incoming SIP messages on an individual dial peer, use the no voice-class sip block command in dial peer voice configuration mode.
NoteThis command is supported only on outbound dial peers--it is nonoperational if configured on inbound dial peers. You should configure this command on the outbound SIP leg that sends out the initial INVITE message. Additionally, this feature applies only to SIP-to-SIP calls and will have no effect on H.323-to-SIP calls.
Examples
The following example shows how to configure dial peer 1 to override any global configurations and drop specified incoming SIP provisional response messages regardless whether SDP is present:
Router> enable Router# configure terminal Router(config)# dial-peer voice 1 voip Router(config-dial-peer)# voice-class sip block 181The following example shows how to configure dial peer 1 to override any global configurations and drop specified incoming SIP provisional response messages only if SDP is present:
Router> enable Router# configure terminal Router(config)# dial-peer voice 1 voip Router(config-dial-peer)# voice-class sip block 183 sdp presentThe following example shows how to configure dial peer 1 to override any global configurations and drop incoming SIP provisional response messages only when SDP is not present:
Router> enable Router# configure terminal Router(config)# dial-peer voice 1 voip Router(config-dial-peer)# voice-class sip block 180 sdp absentThe following example shows how to configure a dial peer to use the global configuration settings for dropping incoming SIP provisional response messages:
Router> enable Router# configure terminal Router(config)# dial-peer voice 1 voip Router(config-dial-peer)# voice-class sip block 181 systemThe following example shows how to configure a dial peer to pass all incoming SIP provisional response messages regardless of global configuration settings:
Router> enable Router# configure terminal Router(config)# dial-peer voice 1 voip Router(config-dial-peer)# no voice-class sip block 180Related Commands
Command
Description
block
Configures global configuration for dropping specified SIP provisional response messages on a Cisco IOS voice gateway or Cisco UBE.
map resp-code
Configures global settings on a Cisco UBE for mapping specific incoming SIP provisional response messages to a different SIP response message.
voice-class sip map resp-code
Configures a specific dial peer on a Cisco UBE to map specific incoming SIP provisional response messages to a different SIP response message.
voice-class sip call-route
To enable call routing based on the Destination-Route-String, P-called-party-id and History-Info header values at the dial-peer configuration level, use the voice-class sip call-route command in dial peer voice configuration mode. To disable Header-based routing, use the no form of this command.
voice-class sip call-route { dest-route-string | p-called-party-id | history-info | url } [system]
no voice-class sip call-route { dest-route-string | p-called-party-id | history-info | url }
Syntax Description
dest-route-string
Enables call routing based on the Destination-Route-String.
p-called-party-id
Enables call routing based on the P-Called-Party-Id header.
history-info
Enables call routing based on the History-Info header.
url
Enables call routing based on the URL.
system
(Optional) Uses the global configuration settings to enable call routing based on the header values on this dial peer.
Command Default
Support for call routing based on the Destination-Route-String, P-Called-Party-Id, History-Info headers and URL at the dial peer level is disabled.
Command History
Release
Modification
12.4(22)YB
This command was introduced.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
15.1(2)T
This command was modified. The history-info keyword was added.
15.2(1)T
This command was modified. The url keyword was added.
15.3(3)M
This command was modified. The dest-route-string keyword was added.
Cisco IOS XE Release 3.10S
This command was integrated into Cisco IOS XE Release 3.10S.
Usage Guidelines
Use the voice-class sip call-route command on the inbound dial peer to enable the gateway to route calls based on the received header in a received INVITE message.
The voice-class sip call-route command takes precedence over the call-route command in voice service VoIP SIP configuration mode. However, if the voice-class sip call-route command is used with the system keyword, the gateway uses the settings configured globally by the call-route command.
If multiple call routes are configured, call routing enabled based on destination route string takes precedence over other header configurations. Destination route string configuration is applicable only for outbound dial-peer matching.
Examples
The following example shows how to enable call routing based on the Destination-Route-String, P-Called-Party-Id, History-Info header values and URL at the dial peer configuration level:
Device> enable Device# configure terminal Device(config)# dial-peer voice 2611 voip Device(config-dial-peer)# voice-class sip call-route dest-route-string Device(config-dial-peer)# voice-class sip call-route p-called-party-id Device(config-dial-peer)# voice-class sip call-route history-info Device(config-dial-peer)# voice-class sip call-route urlvoice-class sip calltype-video
To configure the bearer capability setting on an H.320 dial peer so that it supports unrestricted digital media, use the voice-class sip calltype-video command in dial peer voice configuration mode. To return the bearer capability setting for an H.320 dial peer to the default, use the no form of this command.
Command Default
Bearer capability setting for support of unrestricted digital media support is disabled.
voice-class sip copy-list
To configure a list of entities to be sent to the peer call leg on a dial peer, use the voice-class sip copy-listcommand in dial peer configuration mode. To disable the configuration, use the no form of this command.
Usage Guidelines
Use the voice-class sip copy-list command to configure Cisco Unified Border Element (UBE) to pass an unsupported parameter present in a mandatory header from one peer call leg to another. You can copy the inbound message headers into variables and pass the headers to the outbound peer call leg.
voice-class sip e911
To enable SIP E911 system services on a dial peer, use the voice-class sip e911command in VoIP dialpeer configuration mode. To disable SIP E911 services, use the no form of this command.
Usage Guidelines
The no form of this command sets the dial peer configuration to disable, which indicates that E911 will not be used for this peer. Because the no version of the command causes non default behavior, it can been seen in the show running-config output. See also the voice service voip sip e911 and debug csm neat commands.
Examples
The following examples enable and disable E911 services on a VoIP dial peer:
Router(config)# dial-peer voice 2 Router(config-dial-peer)# voice-class sip e911 *Jun 06 00:47:20.611: setting peer 2 to enable Router(config-dial-peer)# no voice-class sip e911 *Jun 06 00:49:58.931: setting peer 2 to disablevoice-class sip encap clear-channel
To enable RFC 4040-based clear-channel codec negotiation for Session Initiation Protocol (SIP) calls on an individual dial peer, overriding the global setting on a Cisco IOS voice gateway or Cisco Unified Border Element (Cisco UBE), use the voice-class sip encap clear-channel command in dial peer voice configuration mode. To disable RFC 4040-based clear-channel codec negotiation on an individual dial peer for SIP calls on a Cisco IOS voice gateway or Cisco UBE, use the no form of this command.
voice-class sip encap clear-channel [ standard | system ]
no voice-class sip encap clear-channel standard
Command Default
The dial peer uses the system configuration. (If the global encap clear-channel standard command is not enabled, then legacy encapsulation [X-CCD/8000] is used for clear-channel codec negotiation.)
Usage Guidelines
Use the voice-class sip encap clear-channel standard command in dial peer voice configuration mode to override global settings for clear-channel codec negotiation on a Cisco IOS voice gateway or Cisco UBE and enable RFC 4040-based clear-channel codec negotiation [CLEARMODE/8000] for SIP calls on a specific dial peer. RFC 4040-based clear-channel codec negotiation allows dial peers on Cisco IOS voice gateways and Cisco UBEs to successfully interoperate with third-party SIP gateways that do not support legacy Cisco IOS clear-channel codec encapsulation [X-CCD/8000].
When the voice-class sip encap clear-channel standard command is enabled on a specific dial peer on a Cisco IOS voice gateway or Cisco UBE, SIP calls on that dial peer that use the Cisco IOS clear channel codec are translated into calls that use [CLEARMODE/8000] regardless of the global configuration so that the calls do not get rejected when they reach third-party SIP gateways.
You can also use the voice-class sip encap clear-channel system command to configure a specific dial peer to use global configuration settings for clear-channel codec negotiation. To enable RFC 4040 clear-channel codec negotiation for SIP calls globally on a Cisco IOS voice gateway or Cisco UBE, use the encap clear-channel standard command in voice service SIP configuration mode. To override global settings and disable RFC 4040-based clear-channel codec negotiation on a specific dial peer, use the no voice-class sip encap clear-channel standard command in dial peer voice configuration mode.
Examples
The following example shows how to configure dial peer 1 to override any global configurations and enable RFC 4040-based clear-channel codec negotiation for SIP calls:
Router> enable Router# configure terminal Router(config)# dial-peer voice 1 voip Router(config-dial-peer)# voice-class sip encap clear-channel standardThe following example shows how to configure dial peer 1 to use the global configuration for clear-channel codec negotiation for SIP calls:
Router> enable Router# configure terminal Router(config)# dial-peer voice 1 voip Router(config-dial-peer)# voice-class sip encap clear-channel system
