- A
- B
- cac master through call application stats
- call application voice through call denial
- call fallback through called-number (dial peer)
- caller-id (dial peer) through ccm-manager switchover-to-backup
- ccs connect (controller) through clear vsp statistics
- clid through credentials (sip-ua)
- default (auto-config application) through direct-inward-dial
- disable-early-media through dualtone
- E
- F
- G
- H
- icpif through irq global-request
- isdn bind-l3 through ixi transport http
- K
- L
- map q850-cause through mgcp package-capability
- mgcp persistent through mmoip aaa send-id secondary
- mode (ATM/T1/E1 controller) through mwi-server
- N
- O
- package through pattern
- periodic-report interval through proxy h323
- Q
- R
- sccp through service-type call-check
- session through sgcp tse payload
- show aal2 profile through show call filter match-list
- show call history fax through show debug condition
- show dial-peer through show gatekeeper zone prefix
- show gateway through show modem relay statistics
- show mrcp client session active through show sip dhcp
- show sip service through show trunk hdlc
- show vdev through show voice statistics memory-usage
- show voice trace through shutdown (voice-port)
- signal through srv version
- ss7 mtp2-variant through switchover method
- target carrier-id through timeout tsmax
- timeouts call-disconnect through timing clear-wait
- timing delay-duration through type (voice)
- U
- vad (dial peer) through voice-class sip encap clear-channel
- voice-class sip error-code-override through vxml version 2.0
- W
- Z
- disable-early-media 180
- disc_pi_off
- disconnect-ack
- dnis (DNIS group)
- dnis-map
- domain-name (annex G)
- drop-last-conferee
- ds0 busyout (voice)
- ds0-group (E1)
- ds0-group (T1)
- ds0-num
- dsn
- dsp allocation signaling dspid
- dsp services dspfarm
- dspfarm (DSP farm)
- dspfarm (voice-card)
- dspfarm confbridge maximum sessions
- dspfarm connection interval
- dspfarm profile
- dspfarm rtp timeout
- dspfarm transcoder maximum sessions
- dspint dspfarm
- dtmf-interworking rtp-nte
- dtmf timer inter-digit
- dtmf-relay (Voice over Frame Relay)
- dtmf-relay (Voice over IP)
- dualtone
disable-early-media 180
To specify which call treatment, early media or local ringback, is provided for 180 responses with 180 responses with Session Description Protocol (SDP), use the disable-early-media 180 command in sip-ua configuration mode. To enable early media cut-through for 180 messages with SDP, use the no form of this command.
disable-early-media 180
no disable-early-media 180
Syntax Description
This command has no arguments or keywords.
Command Default
Early media cut-through for 180 responses with SDP is enabled.
Command Modes
SIP-UA configuration
Command History
|
|
|
|---|---|
12.2(13)T |
This command was introduced. |
IOS Release XE 2.5 |
This command was integrated into Cisco IOS XE Release 2.5. |
Usage Guidelines
This command provides the ability to enable or disable early media cut-through on Cisco IOS gateways for Session Initiation Protocol (SIP) 180 responses with SDP. Use the disable-early-media 180 command to configure the gateway to ignore the SDP message and provide local ringback. To restore the default treatment, early media cut-through, use the no disable-early-media 180 command.
Examples
The following example disables early media cut-through for SIP 180 responses with SDP:
Router(config-sip-ua)# disable-early-media 180
Related Commands
disc_pi_off
To enable an H.323 gateway to disconnect a call when it receives a disconnect message with a progress indicator (PI) value, use the disc_pi_off command in voice-port configuration mode. To restore the default state, use the no form of this command.
disc_pi_off
no disc_pi_off
Syntax Description
This command has no arguments or keywords.
Command Default
The gateway does not disconnect a call when it receives a disconnect message with a PI value.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The disc_pi_off voice-port command is valid only if the disconnect with PI is received on the inbound call leg. For example, if this command is enabled on the voice port of the originating gateway, and a disconnect message with PI is received from the terminating switch, the disconnect message is converted to a disconnect message. But if this command is enabled on the voice port of the terminating gateway, and a disconnect message with PI is received from the terminating switch, the disconnect message is not converted to a standard disconnect message because the disconnect message is received on the outbound call leg.
Note 
The disc_pi_off voice-port configuration command is valid only for the default session application; it does not work for interactive voice response (IVR) applications.
Examples
The following example handles a disconnect message with a PI value in the same way as a standard disconnect message for voice port 0:23:
voice-port 0:D
disc_pi_off
Related Commands
|
|
|
|---|---|
isdn t306 |
Sets a timer for Disconnect messages. |
disconnect-ack
To configure a Foreign Exchange Station (FXS) voice port to return an acknowledgment upon receipt of a disconnect signal, use the disconnect-ack command in voice-port configuration mode. To disable the acknowledgment, use the no form of this command.
disconnect-ack
no disconnect-ack
Syntax Description
This command has no arguments or keywords.
Command Default
FXS voice ports return an acknowledgment upon receipt of a disconnect signal
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The disconnect-ack command configures an FXS voice port to remove line power if the equipment on an FXS loop-start trunk disconnects first.
Examples
The following example, which begins in global configuration mode, turns off the disconnect acknowledgment signal on voice port 1/1/0:
voice-port 1/0/0
no disconnect-ack
Command History
|
|
|
show voice port |
Displays voice port configuration information. |
dnis (DNIS group)
To add a dialed number identification service (DNIS) number to a DNIS map, use the dnis command in DNIS-map configuration mode. To delete a DNIS number, use the no form of this command.
dnis number [url url]
no dnis
Syntax Description
Command Default
If no URL is entered, the DNIS number links to the VoiceXML application that is configured in the dial peer with the application command.
Command Modes
DNIS-map configuration
Command History
|
|
|
|---|---|
12.2(2)XB |
This command was introduced on the Cisco AS5300, Cisco AS5350, and Cisco AS5400. |
12.2(11)T |
This command was implemented on the Cisco 3640 and Cisco 3660. |
Usage Guidelines
To enter DNIS-map configuration mode for the dnis command, use the voice dnis-map command.
Enter the dnis command once for each telephone number that you want to map to a voice application. A separate entry must be made for each telephone number in a DNIS map. Wildcards are not supported.
URLs in DNIS entries are used only by VoiceXML applications. When an incoming called number matches a DNIS entry, it loads the VoiceXML document that is specified by the URL, provided that a VoiceXML application is configured in the dial peer using the application command.
Non-VoiceXML applications, such as TCL applications, ignore the URLs in DNIS maps and link a call to the TCL application that is configured in the dial peer using the application command.
For a DNIS map to be applied to an outbound dial peer, a VoiceXML application must be configured by using the application command with the out-bound keyword. Otherwise, the call is not handed off to the application that is specified in the URL of the DNIS map.
The number of allowable DNIS entries is limited by the amount of available configuration memory on the gateway. As a guideline, DNIS maps that contain more than several hundred DNIS entries should be maintained in an external text file.
To associate a DNIS map with a dial peer, use the dnis-map command.
Examples
The first line in the following example shows how the voice dnis-map command is used to create a DNIS map named "dmap1". The last two lines show how the dnis command is used to enter DNIS entries.
The first DNIS entry specifies the location of a VoiceXML document. The second DNIS entry does not specify a URL. A DNIS number without a URL is, by default, matched to the URL of the application that is configured in the dial peer by using the application command.
voice dnis-map dmap1
dnis 5553305 url tftp://blue/sky/test.vxml
dnis 5558888
Related Commands
dnis-map
To associate a dialed number identification service (DNIS) map with a dial peer, use the dnis-map command in dial peer configuration mode. To remove a DNIS map from the dial peer, use the no form of this command.
dnis-map map-name
no dnis-map
Syntax Description
map-name |
Name of the configured DNIS map. |
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Command History
|
|
|
|---|---|
12.2(2)XB |
This command was introduced on the Cisco AS5300, Cisco AS5350, and Cisco AS5400. |
12.2(11)T |
This command was implemented on the Cisco 3640 and Cisco 3660. |
Usage Guidelines
A DNIS map is a table of destination numbers with optional URLs that link to specific VoiceXML documents. When configured in a dial peer, a DNIS map enables you to link multiple called numbers to a single Tool Command Language (TCL) application or to individual VoiceXML documents.
The dnis-map command must be used with the application command.
Only one DNIS map can be configured in each dial peer.
To create a DNIS map, use the voice dnis-map command to enter DNIS-map configuration mode, and then use the dnis command to add entries to the DNIS map. Or you can create an external text file of DNIS entries and link to its URL by using the voice dnis-map command.
To view the configuration information for DNIS maps, use the show voice dnis-map command.
A URL configured for a DNIS number is ignored by a TCL application; the TCL script that is configured for the application is used instead.
Note For a DNIS map to be applied to an outbound dial peer, the call application must be configured as an outbound application. That is, a VoiceXML application must be configured by using the application command with the out-bound keyword. Otherwise, the call is not handed off to the application that is specified in the URL of the DNIS map.
Examples
In the following example the DNIS map named "dmap1" is associated with the VoIP dial peer 3. The outbound application "vapptest1" is associated through this dial peer with DNIS map "dmap1".
dial-peer voice 3 voip
dnis-map dmap1
application vapptest1 outbound
Related Commands
domain-name (annex G)
To set the domain name that is reported in service relationships, use the domain-name command in annex G neighbor configuration mode. To remove the domain name, use the no form of this command.
domain-name id
no domain-name id
Syntax Description
id |
Domain name that is reported in service relationships. |
Command Modes
Annex G neighbor configuration mode
Command Default
No default behavior or values
Command History
|
|
|
|---|---|
12.2(11)T |
This command was introduced. |
Usage Guidelines
Use this command to set the domain name reported that is reported in service relationships.
Examples
The following example shows how to set a domain name to "boston1":
Router(config-annexg-neigh)# domain-name boston1
Related Commands
|
|
|
|---|---|
access-policy |
Requires that a neighbor be explicitly configured. |
drop-last-conferee
To define a Feature Access Code (FAC) to access the Drop Last Conferee feature in feature mode on analog phones controlled by Cisco Unified Communications Manager Express (CME), use the drop-last-conferee command in STC application feature-mode call-control configuration mode. To return the code to its default, use the no form of this command.
drop-last-conferee keypad-character
no drop-last-conferee
Syntax Description
keypad-character |
Character string of one to four characters that can be dialed on a telephone keypad (0—9, *, #). Default is #4. |
Command Default
The default value is #4.
Command Modes
STC application feature-mode call-control configuration (config-stcapp-fmcode)
Command History
|
|
|
|---|---|
15.0(1)M |
This command was introduced. |
Usage Guidelines
This command changes the value of the FAC for the Drop Last Conferee feature from the default (#4) to the specified value.
If you attempt to configure this command with a value that is already configured for another FAC in feature mode, you receive a message. This message will not prevent you from configuring the feature code. If you configure a duplicate FAC, the system implements the first feature it matches in the order of precedence as determined by the value for each FAC (#1 to #5).
If you attempt to configure this command with a value that precludes or is precluded by another FAC in feature mode, you receive a message. If you configure a FAC to a value that precludes or is precluded by another FAC in feature mode, the system always executes the call feature with the shortest code and ignores the longer code. For example, 1 will always preclude 12 and 123. These messages will not prevent you from configuring the feature code. You must configure a new value for the precluded code in order to enable phone user access to that feature.
Note This command does not change the user experience for Drop Last Conferee if the Cisco call-control system is Cisco Unified Communications Manager.
Examples
The following example shows how to change the value of the feature code for the Drop Last Conferee feature from the default (#4). With this configuration, a phone user in a three-party conference on an analog phone controlled by Cisco Unified CME presses hook flash to get the feature tone and then dials 44 to drop the last active party. The conference becomes a basic call to the second call party.
Router(config)# stcapp call-control mode feature
Router(config-stcapp-fmcode)# drop-last-conferee 44
Router(config-stcapp-fmcode)# exit
Related Commands
ds0 busyout (voice)
To force a DS0 time slot on a controller into the busyout state, use the ds0 busyout command in controller configuration mode. To remove the DS0 time slot from the busyout state, use the no form of this command.
ds0 busyout ds0-time-slot
no ds0 busyout ds0-time-slot
Syntax Description
ds0-time-slot |
DS0 time slots to be forced into the busyout state. Range is from 1 to 24 and can include any combination of time slots. |
Command Default
DS0 time slots are not in busyout state.
Command Modes
Controller configuration
Command History
|
|
|
|---|---|
12.0(7)XK |
This command was introduced on Cisco MC3810 and Cisco 2600 series and Cisco 3600 series. |
12.1(2)T |
This command was integrated into Cisco IOS Release 12.1(2)T. |
Usage Guidelines
The ds0 busyout command affects only DS0 time slots that are configured into a DS0 group and that function as part of a digital voice port. If multiple DS0 groups are configured on a controller, any combination of DS0 time slots can be busied out, provided that each DS0 time slot to be busied out is part of a DS0 group.
If a DS0 time slot is in the busyout state, only the no ds0 busyout command can restore the DS0 time slot to service.
To avoid conflicting command-line interface (CLI) commands, do not use the ds0 busyout command and the busyout forced command on the same controller.
Examples
The following example configures DS0 time slot 6 on controller T1 0 to be forced into the busyout state:
controller t1 0
ds0 busyout 6
The following example configures DS0 time slots 1, 3, 4, 5, 6, and 24 on controller E1 1 to be forced into the busyout state:
controller e1 1
ds0 busyout 1,3-6,24
Related Commands
|
|
|
busyout seize |
Changes the busyout seize procedure for a voice port. |
show running configuration |
Determines which DS0 time slots have been forced into the busyout state. |
ds0-group (E1)
To specify the DS0 time slots that make up a logical voice port on an E1 controller, specify the signaling type by which the router communicates with the PBX or PSTN, and define E1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, use the ds0-group command in controller configuration mode. To remove the group and signaling setting, use the no form of this command.
Cisco IOS Release 12.2 and Later Releases
Cisco 1750 and Cisco 1751
ds0-group ds0-group-number timeslots timeslot-list {[service service-type] | [type e&m-fgb | e&m-fgd | e&m-immediate-start | fgd-eana | fgd-os | fxs-ground-start | fxs-loop-start | none | r1-itu | r1-modified | r1-turkey]}
no ds0-group ds0-group-number
Cisco IOS Release 12.1 and Earlier Releases
Cisco 1750 and Cisco 1751
ds0-group ds0-group-number timeslots timeslot-list {[service service-type] | [type e&m-fgb | e&m-fgd | e&m-immediate-start | fgd-eana | fgd-os | fxs-ground-start | fxs-loop-start | none | r1-itu | r1-modified | r1-turkey | sas-ground-start | sas-loop-start]}
no ds0-group ds0-group-number
Cisco 2600 Series (Except Cisco 2691), Cisco 3600 Series (Except Cisco 3660)
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start | e&m-melcas-delay | e&m-melcas-immed | e&m-melcas-wink | e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start | fxo-melcas | fxs-ground-start | fxs-loop-start | fxs-melcas | r2-analog | r2-digital | r2-pulse}
no ds0-group ds0-group-number
Cisco 2691, Cisco 2600XM Series, Cisco 2800 Series (Except Cisco 2801), Cisco 3660, Cisco 3700 Series, Cisco 3800 Series
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start | e&m-lmr | e&m-melcas-delay | e&m-melcas-immed | e&m-melcas-wink | e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start | fxo-melcas | fxs-ground-start | fxs-loop-start | fxs-melcas | r2-analog | r2-digital | r2-pulse}
no ds0-group ds0-group-number
Cisco 7200 Series and Cisco 7500 Series Voice Ports
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-wink-start | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start }
no ds0-group ds0-group-number
Cisco 7700 Series Voice Ports
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start | e&m-wink-start | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}
no ds0-group ds0-group-number
Cisco AS5300 and the Cisco AS5400
ds0-group ds0-group-number timeslots timeslot-list type {none | p7 | r2-analog | r2-digital | r2-lsv181-digital | r2-pulse}
no ds0-group ds0-group-number
Note This command does not support the extended echo canceller (EC) feature on the Cisco AS5x00 series.
Syntax Description
Command Default
There is no DS0 group. Calls are allowed in both directions.
Command Modes
Controller configuration
Command History
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows:
•Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, and Cisco 3745, and Cisco 7200 series:
–slot/port:ds0-group-number
Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Be sure you take the following into account when you are configuring DS0 groups:
•Channel groups, CAS voice groups, DS0 groups, and time-division multiplexing (TDM) groups all use group numbers. All group numbers configured for channel groups, CAS voice groups, DS0 groups, and TDM groups must be unique on the local router. For example, you cannot use the same group number for a channel group and for a TDM group.
•The keywords available for the ds0-group command are dependent upon the Cisco IOS software release that you are using. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
http://www.cisco.com/go/fn
•When you are using command-line interface (CLI) help, the keywords for the ds0-group command are configuration specific. For example, if Media Gateway Control Protocol (MGCP) is configured, you see the mgcp keyword. If you are not using MGCP, you do not see the mgcp keyword.
•Cisco IOS Releases later than 12.2 do not support the Single Attachment Station (SAS) CAS options of sas-loop-start and sas-ground-start.
Examples
The following example shows ranges of E1 controller time slots configured for FXS ground-start and FXO loop-start signaling:
E1 1/0
framing esf
linecode b8zs
ds0-group 1 timeslots 1-10 type fxs-ground-start
ds0-group 2 timeslots 11-24 type fxo-loop-start
The following example shows ranges of T1 controller time slots configured for FXS ground-start signaling:
controller E1 1/0
ds0-group 1 timeslots 1-4 type fxs-ground-start
The following example illustrates setting the E1 channels for Signaling System 7 (SS7) service on any trunking gateway using the mgcp keyword:
Router(config-controller)# ds0-group 0 timeslots 1-24 type none service mgcp
In the following example, the time slot maximum is 12 and the time slot is 1, so two voice-ports are created successfully.
controller E1 0/0
ds0-group 0 timeslots 1-4 type e&m-immediate-start
ds0-group 1 timeslots 6-12 type e&m-immediate-start
If a third DS0 group is added, the voice-port is rejected even though the total number of voice channels is less than 16.
ds0-group 2 timeslots 17-18 type e&m-immediate-start
In the following example, the signaling type is set to e&m-lmr:
ds0-group 0 timeslots 1-10 type e&m-lmr
Related Commands
ds0-group (T1)
To specify the DS0 time slots that make up a logical voice port on a T1 controller, to specify the signaling type by which the router communicates with the PBX or PSTN, and to define T1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, use the ds0-group command in controller configuration mode. To remove the group and signaling setting, use the no form of this command.
Cisco IOS Release 12.2 and Later Releases
Cisco 1750 and Cisco 1751
ds0-group ds0-group-number timeslots timeslot-list [service service-type] type {e&m-fgb | e&m-fgd | e&m-immediate-start | fgd-eana | fgd-os | fxs-ground-start | fxs-loop-start | none | r1-itu | r1-modified | r1-turkey}
no ds0-group ds0-group-number
Cisco IOS Release 12.1 and Earlier Releases
Cisco 1750 and Cisco 1751
ds0-group ds0-group-number timeslots timeslot-list [service service-type] type {e&m-fgb | e&m-fgd | e&m-immediate-start | fgd-eana | fgd-os | fxs-ground-start | fxs-loop-start | none | r1-itu | r1-modified | r1-turkey | sas-ground-start | sas-loop-start}
no ds0-group ds0-group-number
Cisco 2600 Series (Except Cisco 2691), Cisco 3600 Series (Except Cisco 3660), and Cisco VG 200
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start}
no ds0-group ds0-group-number
Cisco 2691, Cisco 2600XM Series, Cisco 2800 Series (Except Cisco 2801), Cisco 3660, Cisco 3700 Series, Cisco 3800 Series
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-lmr | e&m-wink-start | ext-sig | fgd-eana | fgd-emf [mf] [ani-pani] [ani] | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start}
no ds0-group ds0-group-number
Cisco 7200 Series and Cisco 7500 Series
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-wink-start | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start}
no ds0-group ds0-group-number
Cisco 7700 Series Voice Ports
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start | e&m-wink-start | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start}
no ds0-group ds0-group-number
Cisco IOS Release 12.2 and Later Releases
Cisco AS5300, Cisco AS5350, and Cisco AS5400
ds0-group ds0-group-number timeslots timeslot-list [service service-type] [type [e&m-fgb [dtmf | mf] | e&m-fgd [dtmf | mf [dnis | ani-dnis [info-digits-no-strip] | | fgd-emf [ani-pani] [ani] | service service-type] | e&m-immediate-start | fxs-ground-start | fxs-loop-start | fgd-eana [ani-dnis | mf] | fgd-os [dnis-ani | mf]| none]]
no ds0-group ds0-group-number
Cisco AS5850
ds0-group ds0-group-number timeslots timeslot-list [service service-type] [type [e&m-fgb [dtmf | mf] | e&m-fgd [dtmf | mf [dnis | ani-dnis [info-digits-no-strip] | | fgd-emf [ani-pani] [ani] | service service-type] | e&m-immediate-start | fxs-ground-start | fxs-loop-start | fgd-eana [ani-dnis | mf] | fgd-os [dnis-ani | mf] | r1-itu [dnis] | none]]
no ds0-group ds0-group-number
Cisco IOS Release 12.1 and Earlier Releases
Cisco AS5300, Cisco AS5350, and Cisco AS5400
ds0-group ds0-group-number timeslots timeslot-list [service service-type] [type [e&m-fgb [dtmf | mf] | e&m-fgd [dtmf | mf [dnis | ani-dnis [info-digits-no-strip] | | fgd-emf [ani-pani] [ani] | service service-type] | e&m-immediate-start | fxs-ground-start | fxs-loop-start | fgd-eana [ani-dnis | mf] | fgd-os [dnis-ani | mf] | sas-ground-start | sas-loop-start | none]]
no ds0-group ds0-group-number
Cisco AS5850
ds0-group ds0-group-number timeslots timeslot-list [service service-type] [type [e&m-fgb [dtmf | mf] | e&m-fgd [dtmf | mf [dnis | ani-dnis [info-digits-no-strip] | | fgd-emf [ani-pani] [ani] | service service-type] | e&m-immediate-start | fxs-ground-start | fxs-loop-start | fgd-eana [ani-dnis | mf] | fgd-os [dnis-ani | mf] | r1-itu [dnis] | sas-ground-start | sas-loop-start | none]]
no ds0-group ds0-group-number
Syntax Description
ds0-group-number |
A value that identifies the DS0 group. Range is from 0 to 23. |
timeslots timeslot-list |
Lists time slots in the DS0 group. The timeslot-list argument is a single time-slot number, a single range of numbers, or multiple ranges of numbers separated by commas. Range is from 1 to 24. Examples are as follows: • • • • |
• |
Specifies the type of signaling for the DS0 group. The signaling method selection for the type keyword depends on the connection that you are making. The ear and mouth (E&M) interface allows connection for PBX trunk lines (tie lines) and telephone equipment. The Foreign Exchange Station (FXS) interface allows connection of basic telephone equipment and PBX. The Foreign Exchange Office (FXO) interface is for connecting the central office (CO) to a standard PBX interface where permitted by local regulations; it is often used for off-premise extensions (OPXs). Types are as follows: • • • • • • • • • • • • • • • • • |
• • • |
|
service service-type |
(Optional) Specifies the type of service. • • • • • |
dtmf |
(Optional) Specifies dual tone multifrequency (DTMF) tone signaling. |
mf |
(Optional) Specifies multifrequency (MF) tone signaling |
ani |
(Optional) Provisions ANI address information. |
ani-dnis |
(Optional) Specifies automatic number identification (ANI) and dialed number identification service (DNIS) address information provisioning for FGD OS. |
ani-pani |
(Optional) Provisions ANI and PANI address information. |
dnis-ani |
(Optional) Specifies ANI and DNIS address information provisioning for FGD EANA. |
dnis |
(Optional) Specifies DNIS address information provisioning. |
info-digits-no-strip |
(Optional) Retains info digits on the Cisco AS5x00 platforms. |
1 Used only with the type none keywords on the Cisco AS5x00 platforms. |
Command Default
There is no DS0 group. Calls are allowed in both directions.
Command Modes
Controller configuration
Command History
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows:
•Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, Cisco 3745, and Cisco 7200 series:
–slot/port:ds0-group-number
•Cisco AS5300, Cisco AS5350, and Cisco AS5400 with a T1 controller:
–slot/port
•Cisco AS5850 with a T1 controller:
–slot/port:ds0-group-number
Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Be sure that you take the following into account when you are configuring DS0 groups:
•Channel groups, CAS voice groups, DS0 groups, and time-division multiplexing (TDM) groups all use group numbers. All group numbers configured for channel groups, CAS voice groups, DS0 groups, and TDM groups must be unique on the local router. For example, you cannot use the same group number for a channel group and for a TDM group.
•The keywords available for the ds0-group command are dependent upon the Cisco IOS software release that you are using. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
•When you are using command-line interface (CLI) help, the keywords for the ds0-group command are configuration specific. For example, if Media Gateway Control Protocol (MGCP) is configured, you see the mgcp keyword. If you are not using MGCP, you do not see the mgcp keyword.
Note This command does not support the extended echo canceller (EC) feature on the Cisco AS5x00 series.
Note The signaling type R1-ITU is not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 platforms.
Examples
The following example shows ranges of T1 controller time slots configured for FXS ground-start and FXO loop-start signaling:
controller T1 1/0
framing esf
linecode b8zs
ds0-group 1 timeslots 1-10 type fxs-ground-start
ds0-group 2 timeslots 11-24 type fxo-loop-start
The following example shows ranges of T1 controller time slots configured for FXS ground-start signaling:
controller T1 1/0
ds0-group 1 timeslots 1-4 type fxs-ground-start
The following example illustrates setting the T1 channels for Signaling System 7 (SS7) service on any trunking gateway using the mgcp keyword:
ds0-group 0 timeslots 1-24 type none service mgcp
In the following example, the time slot maximum is 12 and the time slot is 1, so two voice-ports are created successfully.
controller T1 0/0
ds0-group 0 timeslots 1-4 type e&m-immediate-start
ds0-group 1 timeslots 6-12 type e&m-immediate-start
If a third DS0 group is added, the voice port is rejected even though the total number of voice channels is less than 16.
ds0-group 2 timeslots 17-18 type e&m-immediate-start
In the following example, the signaling type is set to E&M LMR:
ds0-group 0 timeslots 1-10 type e&m-lmr
You have the option to retain info digits when you are configuring E&M Type II Feature Group D with MF signaling and ANI/DNIS for calls being sent over IP. Info digits denote the subscriber type, and the info-digits keyword prepends info digits to the calling number.
On inbound calls from a T1 FGD voice-port with MF ANI-DNIS, when ANI information is obtained, it is passed unaltered to the next matching dial peer, either POTS or VoIP. The addition of the info-digits-no-strip keyword allows you to retain the info digits portion of the ANI information; the modified ANI is then passed to the next matching dial peer. Ordinarily, info digits are not valid for calls going over IP and are, therefore, stripped off. The ability to retain info digits is particularly useful for calls that are not leaving the PSTN network and are just being hairpinned back.
In the following example, the E&M Type II Feature Group D is configured with MF signaling and ANI/DNIS over IP while retaining info digits:
ds0-group 0 timeslots 1-24 type e&m-fgd mf ani-dnis info-digits-no-strip
The following example enables FGD EMF:
ds0-group 11 timeslots 11 type fgd-emf ani
ds0-group 11 timeslots 11 type fgd-emf ani-pani
Related Commands
ds0-num
To add B-channel information in outgoing Session Initiation Protocol (SIP) messages, use the ds0-num command in SIP voice service configuration mode. To return to the default setting, use the no form of this command.
ds0-num
no ds0-num
Syntax Description
This command has no arguments or keywords.
Command Default
B channel information is disabled.
Command Modes
SIP voice service configuration
Command History
|
|
|
|---|---|
12.3(7)T |
This command was introduced. |
Usage Guidelines
This command enables the SIP application to receive B-channel information of incoming ISDN calls. The B-channel information appears in the Via header of an Invite request and information acquired from the Via header can be used during call transfer or to route a call.
Examples
The following example adds B-channel information to outgoing SIP messages:
Router(config)# voice service voip
Router(conf-voi-serv)# sip
Router(conf-serv-sip)# ds0-num
Related Commands
|
|
|
|---|---|
sip |
Enables SIP voice service configuration commands. |
voice service voip |
Specifies the voice encapsulation type as VoIP. |
dsn
To specify that a delivery status notice (DSN) be delivered to the sender, use the dsn command in dial peer configuration mode. To cancel a specific DSN option, use the no form of this command.
dsn {delay | failure | success}
no dsn {delay | failure | success}
Syntax Description
Note In the absence of any other DSN settings (for example, no dsn, or a mailer in the path that does not support the DSN extension), a failure to deliver message always causes a nondelivery message to be generated. This nondelivery message is called a bounce.
Command Default
The default is to send a nondelivery message in the event of a failure
Command Modes
Dial peer configuration
Command History
Usage Guidelines
When the delay keyword is selected, the next-hop mailer sends a message to the FROM address saying that the mail message was delayed. The definition of the delay keyword is made by each mailer and is not controlled by the sender. Each mailer in the path to the recipient that supports the DSN extension receives the same request.
When the failure keyword is selected, the next-hop mailer sends a message to the FROM address that the mail message delivery failed. Each mailer in the path to the recipient that supports the DSN extension receives the same request.
When the success keyword is selected, the next-hop mailer sends a message to the FROM address saying that the mail message was successfully delivered to the recipient. Each mailer in the path to the recipient that supports the DSN extension receives the same request.
This command is applicable to Multimedia Mail over Internet Protocol (MMoIP) dial peers.
DSNs are messages or responses that are automatically generated and sent to the sender or originator of an e-mail message by the Simple Mail Transfer Protocol (SMTP) server, notifying the sender of the status of the e-mail message. Specifications for DSN are described in RFC 1891, RFC 1892, RFC 1893, and RFC 1894.
The on-ramp DSN request is included as part of the fax-mail message sent by the on-ramp gateway when the matching MMoIP dial peer has been configured. The on-ramp DSN response is generated by the SMTP server when the fax-mail message is accepted. The DSN is sent back to the user defined by the mta send mail-from command. The off-ramp DSN is requested by the e-mail client. The DSN response is generated by the SMTP server when it receives a request as part of the fax-mail message.
Note DSNs are generated only if the mail client on the SMTP server is capable of responding to a DSN request.
Because the SMTP server generates the DSNs, you need to configure both mail from: and rcpt to: on the server for the DSN feature to work. For example:
mail from: <user@mail-server.company.com>
rcpt to: <fax=555-1212@company.com> NOTIFY=SUCCESS,FAILURE,DELAY
There are three different states that can be reported back to the sender:
•Delay—Indicates that the message was delayed in being delivered to the recipient or mailbox.
•Success—Indicates that the message was successfully delivered to the recipient or mailbox.
•Failure—Indicates that the SMTP server was unable to deliver the message to the recipient or mailbox.
Because these delivery states are not mutually exclusive, you can configure store-and-forward fax to generate these messages for all or any combination of these events.
DSN messages notify the sender of the status of a particular e-mail message that contains a fax TIFF image. Use the dsn command to specify which notification messages are sent to the user.
The dsn command allows you to select more than one notification option by reissuing the command and specifying a different notification option each time. To discontinue a specific notification option, use the no form of the command for that specific keyword.
If the failure keyword is not included when DSN is configured, the sender receives no notification of message delivery failure. Because a failure is usually significant, care should be taken to always include the failure keyword as part of the dsn command configuration.
This command applies to on-ramp store-and-forward fax functions.
Examples
The following example specifies that a DSN message be returned to the sender when the e-mail message that contains the fax has been successfully delivered to the recipient or if the message that contains the fax has failed to be delivered:
dial-peer voice 10 mmoip
dsn success
dsn failure
Related Commands
dsp allocation signaling dspid
To change the digital signal processor (DSP) selection for signaling channel allocation from the default (DSP weight-based) to the DSP ID number, use the dsp allocation signaling dspid command in voice-card configuration mode. To return to the default behavior, use the no form of this command.
dsp allocation signaling dspid
no dsp allocation signaling dspid
Syntax Description
This command has no arguments or keywords.
Command Default
Selection of a DSP for signaling channel allocation is based on the internal weighted value assigned to the DSPs.
Command Modes
Voice-card configuration (config-voicecard)
Command History
|
|
|
|---|---|
12.4(15)T9 |
This command was introduced. |
Usage Guidelines
The dsp allocation signaling dspid command takes effect only after a reload of the router. The command should be enabled and saved into the startup-config file.
The default signal channel allocation method (by weight) may not be suitable for some network implementations. The default allocation method selects the DSPs based on the DSP weight, and you cannot control the selection of the DSP for specific configuration even if the order of the packet voice data modules (PVDMs) is changed. Enable the dsp allocation signaling dspid command to change the selection order to the DSP ID number. This command is more useful when there is a PVDM2-8 module in the network configuration.
Examples
The following example shows how to change the default for DSP allocation from the DSP weight to the DSP ID number:
voice card 1
dsp allocation signaling dspid
Related Commands
|
|
|
|---|---|
show voice dsp |
Displays the current status or selective statistics of DSP voice channels. |
voice-card |
Enters voice-card configuration mode. |
dsp services dspfarm
To enable digital-signal-processor (DSP) farm services for a particular voice network module, use the dsp services dspfarm command in voice card configuration mode. To disable services, use the no form of this command.
dsp services dspfarm
no dsp services dspfarm
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
Voice-card configuration (config-voicecard)
Command History
|
|
|
|---|---|
12.2(13)T |
This command was introduced. |
Cisco IOS XE |
Support for this command was added on Cisco ASR 1000 Series Routers. |
Usage Guidelines
The router must be equipped with one or more voice network modules that provide DSP resources. DSP resources are used only if this command is configured under the particular voice card.
The number of voice network modules that must be enabled for DSP-farm services depends on the number of DSPs on the module and on the maximum number of transcoding and conferencing sessions configured for the DSP farm.
Note Use this command before enabling DSP-farm services with the dspfarm command for an NM-HDV or NM-HDV-FARM.
Cisco ASR 1000 Series Router
The SPA-DSPs on a Cisco ASR 1000 Series Routers are installed in a subslot on a SIP. Hence, when referring to a SPA-DSP the voice-card command is used.
Examples
The following example enables DSP-farm services on an NM-HDV2 or NM-HD-1V/2V/2VE:
Router(config)# voice-card 2
Router(config-voicecard)# dsp services dspfarm
Router(config-voicecard)# exit
The following example enables DSP-farm services on an NM-HDV or NM-HDV-FARM:
Router(config)# voice-card 2
Router(config-voicecard)# dsp services dspfarm
Router(config-voicecard)# exit
The following example enables DSP-farm services on SPA-DSP for a Cisco ASR 1000 Series Router:
Router(config)# voice-card 1/1
Router(config-voicecard)# dsp services dspfarm
Router(config-voicecard)# exit
Related Commands
dspfarm (DSP farm)
To enable digital-signal-processor (DSP) farm service, use the dspfarm command in global configuration mode. To disable the service, use the no form of this command.
dspfarm
no dspfarm
Syntax Description
This command has no arguments or keywords.
Command Default
DSP-farm service is disabled.
Command Modes
Global configuration
Command History
Usage Guidelines
The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms (NM-HDV-FARMs) to provide DSP resources.
Before enabling DSP-farm services, you must configure the NM-HDV or NM-HDV-FARM on which DSP-farm services are to be enabled using the dsp service dspfarm command. You must also specify the maximum number of transcoding sessions to be supported by the DSP farm using the dspfarm transcoder maximum sessions command.
This command causes the system to download new firmware into the DSPs, start up the required subsystems, and wait for a service request from the transcoding and conferencing applications.
Examples
The following example configures an NM-HDV or NM-HDV-FARM, specifies the maximum number of transcoding sessions, and enables DSP-farm services:
Router# configure terminal
Router(config)# no dspfarm
Router(config)# voice-card 2
Router(config-voicecard)# dsp services dspfarm
Router(config-voicecard)# exit
Router(config)# dspfarm transcoder maximum sessions 15
Router(config)# dspfarm
Related Commands
dspfarm (voice-card)
To add a specified voice card to those participating in a digital signal processor (DSP) resource pool, use the dspfarm command in voice-card configuration mode. To remove the specified card from participation in the DSP resource pool, use the no form of this command.
dspfarm
no dspfarm
Syntax Description
This command has no arguments or keywords.
Command Default
A card participates in the DSP resource pool
Command Modes
Voice-card configuration
Command History
Usage Guidelines
DSP mapping occurs when DSP resources on one AIM or network module are available for processing of voice time-division multiplexing (TDM) streams on a different network module or on a voice/WAN interface card (VWIC). This command is used on Cisco 3660 routers with multiservice interchange (MIX) modules installed or on Cisco 2600 series routers with AIMs installed.
To reach voice-card configuration mode for a particular voice card, from global configuration mode enter the voice-card command and the slot number for the AIM or network module that you want to add to the pool. See the voice-card command reference for details on slot numbering.
The assignment of DSP pool resources to particular TDM streams is based on the order in which the streams are configured with the ds0-group command for T1/E1 channel-associated signaling (CAS) or with the pri-group command for ISDN PRI.
The assignment of DSP pool resources does not occur dynamically during call signaling.
Examples
The following example adds to the DSP resource map the DSP resources on the network module in slot 5 on a Cisco 3660 with a MIX module:
voice-card 5
dspfarm
The following example makes available the DSP resources on an AIM on a modular access router:
voice-card 0
dspfarm
Related Commands
dspfarm confbridge maximum sessions
To specify the maximum number of concurrent conference sessions for which digital-signal-processor (DSP) farm resources should be allocated, use the dspfarm confbridge maximum sessions command in global configuration mode. To reset to the default, use the no form of this command.
dspfarm confbridge maximum sessions number
no dspfarm confbridge maximum sessions
Syntax Description
number |
Number of conference sessions. A single DSP supports 1 conference session with up to 6 participants. |
Command Default
0 sessions
Command Modes
Global configuration
Command History
Usage Guidelines
The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms (NM-HDV-FARMs) to provide DSP resources.
Before using this command, you must disable DSP-farm service using the no dspfarm command.
The maximum number of conference sessions depends upon DSP availability in the DSP farm. A single DSP supports one conference session with up to six participants. However, you may need to allocate additional DSP resources for transcoding to support conferences. If all participants use G.711 or G.729 codecs, you need not allocate any additional DSP resources because transcoding is done in the conferencing DSP.
When you use this command, take into consideration the number of DSPs allocated for transcoding services with the dspfarm transcoder maximum sessions command.
Examples
The following example sets the maximum number of conferencing sessions to 8:
Router# dspfarm confbridge maximum sessions 8
Related CommandsT
dspfarm connection interval
To specify the time interval during which to monitor Real-Time Transport Protocol (RTP) inactivity before deleting an RTP stream, use the dspfarm connection interval command in global configuration mode. To reset to the default, use the no form of this command.
dspfarm connection interval seconds
no dspfarm connection interval
Syntax Description
seconds |
Interval, in seconds, during which to monitor RTP inactivity. Range is from 60 to 10800. Default is 600. |
Command Default
600 seconds
Command Modes
Global configuration
Command History
Usage Guidelines
The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms (NM-HDV-FARMs) to provide digital-signal-processor (DSP) resources.
After each interval, RTP streams are checked for inactivity. If all RTP streams for a particular call are inactive, the RTP timer, as set with the dspfarm rtp timeout command, is started. When the RTP timer expires, the call is deleted.
Examples
The following example sets the connection interval to 60 seconds:
Router(config)# dspfarm connection interval 60
Related Commands
|
|
|
|---|---|
dspfarm rtp timeout |
Specifies the RTP timeout interval used to clear hanging connections. |
dspfarm profile
To enter DSP farm profile configuration mode and define a profile for digital signal processor (DSP) farm services, use the dspfarm profile command in global configuration mode. To delete a disabled profile, use the no form of this command.
Cisco Unified Border Element
dspfarm profile profile-identifier {conference | mtp | transcode} [security]
no dspfarm profile profile-identifier
Cisco Unified Border Element (Enterprise) Cisco ASR 1000 Series Router
dspfarm profile profile-identifier {transcode}
no dspfarm profile profile-identifier
Cisco Integrated Services Routers Generation 2 (Cisco ISR G2)
dspfarm profile profile-identifier {conference [video [homogeneous | heterogeneous | guaranteed-audio ] ] | mtp | transcode [video | universal] } [security]
no dspfarm profile profile-identifier
Syntax Description
Command Default
If this command is not entered, no profiles are defined for the DSP farm services.
Command Modes
Global configuration (config)
Command History
Usage Guidelines
Use this command to create a new profile or delete a disabled profile. After you create a new profile in dspfarm profile configuration mode, use the no shutdown command to enable the profile configuration, allocate resources and associate the profile with the application(s). If the profile cannot be enabled due to lack of resources, the system prompts you with a message "Can not enable the profile due to insufficient resources, resources available to support X sessions; please modify the configuration and retry."
If the DSP farm profile is successfully created, you enter the DSP farm profile configuration mode. You can configure multiple profiles for the same service.
Use the no dspfarm profile command to delete a profile from the system. If the profile is active, you cannot delete it; you must first disable it using the shutdown command. To modify a DSP farm profile, use the shutdown command in dspfarm profile configuration mode before you begin configuration.
The profile identifier uniquely identifies a profile. If the service type and profile identifier are not unique, the user is prompted with a message to choose a different profile identifier.
You must use the security keyword in order to enable secure DSP farm services such as secure transcoding.
Effective with Cisco IOS Releases 15.0(1)M2 and 15.1(1)T, platform support for the Cisco IAD 2430, IAD 2431, IAD 2432, and IAD 2435, and the Cisco VG 202, VG 204, and VG 225 is modified. These platforms are designed as TDM-IP devices and are not expandable to install extra DSP resources. So even though the conference keyword appears in the command syntax, this DSP service is not configurable on these platforms. If you try to configure conferencing on these platforms, the command-line interface displays the following message: "%This platform does not support Conferencing feature."
The transcode keyword also appears in the command syntax, but this DSP service is not available on the Cisco VG 202, VG 204, and VG 224 platforms. If you try to configure transcoding on these platforms, the CLI displays the following message: "%This platform does not support Transcoding feature."
Cisco ASR 1000 Series Router
The support for dspfarm profile command was added on Cisco ASR 1000 Series Router from Cisco IOS XE Release 3.2 and later releases. The command is used to create a dspfarm profile for different services.
Note The secure DSP farm services is always enabled for SPA-DSP on Cisco ASR 1000 Series Router. Only transcode keyword is supported on Cisco ASR 1000 Series Router for Cisco IOS XE Release 3.2s. The conference, media, and security keywords are not supported on Cisco ASR 1000 Series Router for Cisco IOS XE Release 3.2s.
In order to configure a video dspfarm profile, you must set voice-service dsp-reservation to be less than 100 percent.
To enable dspfarm profiles for voice services, you must use the dsp services dspfarm command under the voice-card submode.
Examples
The following example enables DSP farm services profile 20 for conferencing:
Router(config)# dspfarm profile 20 conference
Note the response if the profile is already being used:
Router(config)# dspfarm profile 6 conference
Profile id 6 is being used for service TRANSCODING
please select a different profile id
The following example enables DSP farm services profile 1 for transcoding:
Router(config)# dspfarm profile 1 transcode
Video Conferences
The following example enables DSP farm services profile 99 for homogeneous video. The conference supports four participants under one format (Video codec H.263, qcif resolution, and a frame-rate of 15 f/s).
Router(config)# dspfarm profile 99 conference video homogeneous
Router(config-dspfarm-profile)# codec h263 qcif frame-rate 15
Router(config-dspfarm-profile)# maximum conference-participant 4
Related Commands
dspfarm rtp timeout
To specify the Real-Time Transport Protocol (RTP) timeout interval used to clear hanging connections, use the dspfarm rtp timeout command in global configuration mode. To reset to the default, use the no form of this command.
dspfarm rtp timeout seconds
no dspfarm rtp timeout
Syntax Description
seconds |
RTP timeout interval, in seconds. Range is from 10 to 7200. Default is 1200. |
Command Default
1200 seconds (20 minutes)
Command Modes
Global configuration
Command History
Usage Guidelines
The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms (NM-HDV-FARMs) to provide digital-signal-processor (DSP) resources.
Use this command to set the RTP timeout interval for when the error condition "RTP port unreachable" occurs.
Examples
The following example sets the RTP timeout value to 600 seconds (10 minutes):
Router# dspfarm rtp timeout 600
Related Commands
dspfarm transcoder maximum sessions
To specify the maximum number of transcoding sessions to be supported by the digital-signal-processor (DSP) farm, use the dspfarm transcoder maximum sessions command in global configuration mode. To reset to the default, use the no form of this command.
dspfarm transcoder maximum sessions number
no dspfarm transcoder maximum sessions
Syntax Description
number |
Number of transcoding sessions. |
Command Default
0 sessions
Command Modes
Global configuration
Command History
Usage Guidelines
The router on which this command is used must be equipped with one or more digital T1/E1 packet voice trunk network modules (NM-HDVs) or high-density voice (HDV) transcoding/conferencing DSP farms (NM-HDV-FARMs) to provide DSP resources.
Before using this command, you must disable DSP-farm service using the no dspfarm command.
Use this command in conjunction with the dspfarm confbridge maximum sessions commands.
The maximum number of transcoding sessions depends upon DSP availability in the DSP farm. A single DSP supports four transcoding sessions transmission to and from G.711 and G.729 codecs.
Examples
The following example configures an NM-HDV or NM-HDV-FARM, specifies the maximum number of transcoding sessions, and enables DSP-farm services:
Router# configure terminal
Router(config)# no dspfarm
Router(config)# voice-card 2
Router(config-voicecard)# dsp services dspfarm
Router(config-voicecard)# exit
Router(config)# dspfarm transcoder maximum sessions 15
Router(config)# dspfarm
Related Commands
dspint dspfarm
To enable the digital signal processor (DSP) interface, use the dspint dspfarm command in global configuration mode. This command does not have a no form.
dspint dspfarm slot/port
Syntax Description
slot |
Slot number of the interface. |
port |
Port number of the interface. |
Command Default
Enabled
Command Modes
Global configuration
Command History
Usage Guidelines
DSP mapping occurs when DSP resources on one advanced interface module (AIM) or network module are available for processing of voice time-division multiplexing (TDM) streams on a different network module or on a voice/WAN interface card (VWIC). This command is used on Cisco 3660 routers with multiservice interchange (MIX) modules installed or on Cisco 2600 series routers with AIMs installed.
To reach voice-card configuration mode for a particular voice card, from global configuration mode enter the voice-card command and the slot number for the AIM or network module that you want to add to the pool. See the voice-card command reference for details on slot numbering.
The assignment of DSP pool resources to particular TDM streams is based on the order in which the streams are configured using the ds0-group command for T1/E1 channel-associated signaling (CAS) or using the pri-group command for ISDN PRI.
The assignment of DSP pool resources does not occur dynamically during call signaling.
To disable the interface use the no shutdown command.
Examples
The following example creates a DSP farm interface with a slot number of 1 and a port number of 0.
dspint dspfarm 1/0
To change codec complexity on the Cisco 7200 series, you must enter the following commands:
Router# configure terminal
Router(config)# dspint dspfarm 2/0
Router(config-dspfarm)# codec medium | high ecan-extended
Related Commands
dtmf-interworking rtp-nte
To enable a delay between the dtmf-digit begin and dtmf-digit end events in the RFC 2833 packets sent from Cisco Unified Border Element (Cisco UBE) or Cisco Unified Communications Manager Express (Cisco Unified CME), use the dtmf-interworking rtp-nte command in voice-service or dial-peer configuration mode. To remove the delay amount, use the no form of this command.
dtmf-interworking rtp-nte
no dtmf-interworking rtp-nte
Syntax Descriptionno dtmf-interworking rtp-nte
This command has no arguments or keywords.
Command Default
RFC 2833 packet is sent in a single burst of three dtmf-digit begin events, one duration equaling 50ms, and three dtmf-digit end events with a duration of 100ms.
Command Modes
Voice-service configuration (config-voi-serv)
Dial-peer configuration (config-dial-peer)
Command History
|
|
|
|---|---|
12.4(15)XZ |
This command was introduced. |
12.4(20)T |
This command was integrated into Cisco IOS Release 12.4(20)T. |
Usage Guidelines
If your system is configured for RFC 2833 DTMF interworking and if the remote system cannot handle RFC 2833 packets sent in a single burst, use this command to introduce a delay between the dtmf-digit begin and end events in the RFC 2833 packet.
Examples
The following example shows a delay between the dtmf-digit and events being configured.
Router(config-voi-serv) dtmf-interworking rtp-nte
Related Commands
|
|
|
|---|---|
nte-end-digit-delay |
Specifies length of delay for each digit in dtmf-digit end event. |
keypad-normalize |
Ensures that the delay configured for a dtmf-end event is always honored. |
dtmf timer inter-digit
To configure the dual tone multifrequency (DTMF) interdigit timer for a DS0 group, use the dtmf timer inter-digit command in T1 controller configuration mode. To restore the timer to its default value, use the no form of this command.
dtmf timer inter-digit milliseconds
no dtmf timer inter-digit
Syntax Description
milliseconds |
DTMF interdigit timer duration, in milliseconds. Range is from 250 to 3000. The default is 3000. |
Command Default
3000 milliseconds
Command Modes
T1 controller configuration
Command History
|
|
|
|---|---|
12.1(3)T |
This command was introduced on the Cisco AS5300. |
Usage Guidelines
Use the dtmf timer inter-digit command to specify the duration in milliseconds the router waits to detect the end of DTMF digits. After this period, the router expects no more digits to arrive and establishes the call.
Examples
The following example, beginning in global configuration mode, sets the DTMF interdigit timer value to 250 milliseconds:
controller T1 2
ds0-group 2 timeslots 4-10 type e&m-fgb dtmf dnis
cas-custom 2
dtmf timer inter-digit 250
Related Commands
dtmf-relay (Voice over Frame Relay)
To enable the generation of FRF.11 Annex A frames for a dial peer, use the dtmf-relay command in dial peer configuration mode. To disable the generation of FRF.11 Annex A frames and return to the default handling of dial digits, use the no form of this command.
dtmf-relay
no dtmf-relay
Syntax Description
This command has no arguments or keywords.
Command Default
Disabled
Command Modes
Dial peer configuration
Command History
Usage Guidelines
Cisco recommends that this command be used with low bit-rate codecs.
When dtmf-relay (VoFR) is enabled, the digital signal processor (DSP) generates Annex A frames instead of passing a dual-tone multifrequency (DTMF) tone through the network as a voice sample. For information about the payload format of FRF.11 Annex A frames, see the Cisco IOS Wide-Area Networking Configuration Guide.
Examples
The following example shows how to enable FRF.11 Annex A frames for VoFR dial peer 200, starting from global configuration mode:
dial-peer voice 200 vofr
dtmf-relay
Related Commands
dtmf-relay (Voice over IP)
To specify how an H.323 or Session Initiation Protocol (SIP) gateway relays dual tone multifrequency (DTMF) tones between telephony interfaces and an IP network, use the dtmf-relay command in dial peer voice configuration mode. To remove all signaling options and send the DTMF tones as part of the audio stream, use the no form of this command.
dtmf-relay {[cisco-rtp] [h245-alphanumeric] [h245-signal] [rtp-nte [digit-drop]] [sip-notify]}
no dtmf-relay
Syntax Description
Command Default
DTMF tones are disabled and sent in-band. That is, they are left in the audio stream.
Command Modes
Dial peer voice configuration
Command History
Usage Guidelines
DTMF is the tone generated when you press a button on a touch-tone phone. This tone is compressed at one end of a call; when the tone is decompressed at the other end, it can become distorted, depending on the codec used. The DTMF relay feature transports DTMF tones generated after call establishment out-of-band using either a standard H.323 out-of-band method or a proprietary RTP-based mechanism. For SIP calls, the most appropriate method to transport DTMF tones is RTP-NTE or SIP-NOTIFY.
This command specifies how an H.323 or SIP gateway relays DTMF tones between telephony interfaces and an IP network.
You must include one or more keywords when using this command.
To avoid sending both in-band and out-of band tones to the outgoing leg when sending IP-to-IP gateway calls in-band (rtp-nte) to out-of band (h245-alphanumeric), configure the dtmf-relay command using the rtp-nte and digit-drop keywords on the incoming SIP dial peer. On the H.323 side, and for H.323 to SIP calls, configure this command using either the h245-alphanumeric or h245-signal keyword.
The SIP-NOTIFY method sends NOTIFY messages bidirectionally between the originating and terminating gateways for a DTMF event during a call. If multiple DTMF relay mechanisms are enabled on a SIP dial peer and are negotiated successfully, the SIP-NOTIFY method takes precedence.
SIP NOTIFY messages are advertised in an invite message to the remote end only if the dtmf-relay command is set.
For SIP, the gateway chooses the format according to the following priority:
1. sip-notify (highest priority)
2. rtp-nte
3. None—DTMF sent in-band
The gateway sends DTMF tones only in the format that you specify if the remote device supports it. If the H.323 remote device supports multiple formats, the gateway chooses the format according to the following priority:
1. cisco-rtp (highest priority)
2. h245-signal
3. h245-alphanumeric
4. rtp-nte
5. None—DTMF sent in-band
The principal advantage of the dtmf-relay command is that it sends DTMF tones with greater fidelity than is possible in-band for most low-bandwidth codecs, such as G.729 and G.723. Without the use of DTMF relay, calls established with low-bandwidth codecs may have trouble accessing automated DTMF-based systems, such as voice mail, menu-based Automatic Call Distributor (ACD) systems, and automated banking systems.
Note•The cisco-rtp keyword supports a proprietary Cisco implementation and operates only between two Cisco 2600 series or Cisco 3600 series routers running Cisco IOS Release 12.0(2)XH or later. Otherwise, the DTMF relay feature does not function, and the gateway sends DTMF tones in-band.
•The cisco-rtp keyword is supported on Cisco 7200 series routers.
•The sip-notify keyword is available only if the VoIP dial peer is configured for SIP.
•The digit-drop keyword is available only when the rtp-nte keyword is configured.
Examples
The following example configures DTMF relay with the cisco-rtp keyword when DTMF tones are sent to dial peer 103:
dial-peer voice 103 voip
dtmf-relay cisco-rtp
The following example configures DTMF relay with the cisco-rtp and h245-signal keywords when DTMF tones are sent to dial peer 103:
dial-peer voice 103 voip
dtmf-relay cisco-rtp h245-signal
The following example configures the gateway to send DTMF in-band (the default) when DTMF tones to are sent dial peer 103:
dial-peer voice 103 voip
no dtmf-relay
The following example configures DTMF relay with the digit-drop keyword to avoid both in-band and out-of band tones being sent to the outgoing leg on H.323 to H.323 or H.323 to SIP calls:
dial-peer voice 1 voip
session protocol sipv2
dtmf-relay h245-alphanumeric rtp-nte digit-drop
The following example configures DTMF relay with the rtp-nte keyword when DTMF tones are sent to dial peer 103:
dial-peer voice 103 voip
dtmf-relay rtp-nte
The following example configures the gateway to send DTMF tones using SIP NOTIFY messages to dial peer 103:
dial-peer voice 103 voip
session protocol sipv2
dtmf-relay sip-notify
Related Commands
|
|
|
|---|---|
notify telephone-event |
Configures the maximum interval between two consecutive NOTIFY messages for a particular telephone event. |
dualtone
To enter cp-dualtone configuration mode for specifying a custom call-progress tone, use the dualtone command in custom-cptone voice-class configuration mode. To configure the custom-cptone voice class not to detect a call-progress tone, use the no form of this command.
dualtone {busy | conference | disconnect | number-unobtainable | out-of-service | reorder | ringback}
no dualtone {busy | conference | disconnect | number-unobtainable | out-of-service | reorder | ringback}
Syntax Description
Command Default
No call-progress tones are defined within the custom-cptone voice class
Command Modes
Custom-cptone voice-class configuration
Command History
Usage Guidelines
The dualtone command enters cp-dualtone configuration mode and specifies a call-progress tone to be detected. You can specify additional call-progress tones without exiting cp-dualtone configuration mode.
Any call-progress tones that are not specified are not detected.
To delete a call-progress tone from this custom-cptone voice class, use the no form of this command and the keyword for the tone that should not be detected; for example, no dualtone busy.
You must associate the class of custom call-progress tones with a voice port for this command to affect tone detection.
Use the dualtone conference command to define custom join and leave tones for hardware conferences.
Examples
The following example enters cp-dualtone configuration mode and specifies busy tone and ringback tone in the custom-cptone voice class country-x.
Router(config)# voice class custom-cptone country-x
Router(cfg-cptone)# dualtone busy
Router(cfg-cp-dualtone)# frequency 440 480
Router(cfg-cp-dualtone)# cadence 500 500
Router(cfg-cp-dualtone)# exit
Router(cfg-cptone)# dualtone ringback
Router(cfg-cp-dualtone)# frequency 400 440
Router(cfg-cp-dualtone)# cadence 2000 4000
The following example deletes ringback tone from the custom-cptone voice class country-x.
Router(config)# voice class custom-cptone country-x
Router(cfg-cptone)# no dualtone ringback
The following example configures a conference leave tone. The configured leave tone must be associated with a digital signal processor (DSP) farm profile.
Router(config)# voice class custom-cptone leavetone
Router(cfg-cptone)# dualtone conference
Router(cfg-cp-dualtone)# frequency 500 500
Router(cfg-cp-dualtone)# cadence 100 100 100 100 100
Feedback