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disable-early-media through dualtone
- 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
- dscp media
- dscp-profile
- dsn
- dsp allocation signaling dspid
- dsp services dspfarm
- dspfarm (DSP farm)
- dspfarm (voice-card)
- dspfarm confbridge maximum
- dspfarm connection interval
- dspfarm profile
- dspfarm rtp timeout
- dspfarm transcoder maximum sessions
- dspint dspfarm
- dtmf-interworking
- 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.
Usage Guidelines
This command provides the ability to enable or disable early media cut-through on Cisco IOS gateways for 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.
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 noform of this command.
Command Default
The gateway does not disconnect a call when it receives a disconnect message with a PI value.
Command History
Release
Modification
12.1(5)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco 7500 series, Cisco AS5300, Cisco AS5800, and Cisco MC3810.
12.2(2)XA
This command was implemented on the Cisco AS5400 and Cisco AS5350.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into the Cisco IOS Release 12.2(11)T.
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.
NoteThe 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.
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.
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.
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.
Syntax Description
telephone-umber
Adds a user-selected DNIS number to a DNIS map.
url url
(Optional) URL that links a DNIS number to a specific VoiceXML document. If a URL is not entered, the DNIS number is linked to the VoiceXML application in the dial peer, which must be configured using the application command. This keyword is not valid for Tool Command Language (TCL) applications.
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.
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 with the application command configured.
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 with the application out-bound command. 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 general rule, 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 the configured application command.
voice dnis-map dmap1 dnis 5550105 url tftp://blue/sky/test.vxml dnis 5550188dnis-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.
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-mapcommand 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 display 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.
NoteFor 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 with the application out-bound command. 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 outbounddomain-name (annex G)
To set the domain name that is reported in service relationships, use the domain namecommand in annex G neighbor configuration mode. To remove the domain name, use the no form of this command.
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.
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.
NoteThis 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)# exitRelated Commands
Command
Description
conference
Defines FAC in Feature Mode to initiate a three-party conference.
hangup-last-active-call
Defines FAC in feature mode to drop last active call during a three-party conferencee.
toggle-between-two-calls
Defines FAC in feature mode to toggle between two active calls.
transfer
Defines FAC in feature mode to connect a call to a third party that the phone user dials.
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.
Usage Guidelines
The ds0 busyoutcommand 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 interaction of 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 6The 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 1,3-6,24ds0-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 | em-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 | &em-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 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
Syntax Description
Command History
Release
Modification
11.2
This command was introduced for the Cisco AS5300 as the cas-group command.
11.3(1)MA
The command was introduced as the voice-group command for the Cisco MC3810.
12.0(1)T
This command was integrated into Cisco IOS Release 12.0(1)T, and the cas-group command was implemented on the Cisco 3600 series routers.
12.0(5)T
The command was renamed ds0-group on the Cisco AS5300 and Cisco 2600 series and Cisco 3600 series routers. Some keyword modifications were implemented.
12.0(5)XE
This command was implemented on the Cisco 7200 series.
12.0(7)XK
Support for this command was implemented on the Cisco MC3810. When the ds0-group command became available on the Cisco MC3810, the voice-group command was removed and no longer supported. The ext-sig keyword replaced the ext-sig-master and ext-sig-slave keywords that were available with the voice-group command.
12.0(7)XR
The mgcp service type was added.
12.1(2)XH
The e&m-fgd and fgd-eana keywords were added for Feature Group D signaling.
12.1(5)XM
The sgcp keyword was removed.
12.1(3)T
This command was modified for Cisco 7500 series routers. The fgd-os signaling type and the voice service type were added.
12.2
The command was modified to exclude sas keywords. The Single Attachment Station (SAS) CAS options of sas-loop-start and sas-ground-start are not supported as a type of signaling for the DS0 group.
12.2(2)XA
This command was implemented on the Cisco AS5300.
12.2(2)T
This command was integrated into Cisco IOS Release 12.2(2)T and implemented on the Cisco 7200 series.
12.2(4)T
Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on Cisco 1750 and Cisco 1751 routers. Support for other Cisco platforms is not included in this release.
12.2(2)XN
Support for the mgcp keyword was added to Cisco CallManager Version 3.1 for the Cisco 2600 series, Cisco 3600 series, and Cisco VG200.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command was supported with Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2. This command is supported on the Cisco IAD2420 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5850 in this release.
12.2(13)T
This command was integrated into Cisco IOS Release 12.2(13)T. The Cisco 1750 and Cisco 1751 do not support T1 and E1 voice and data cards in Cisco IOS Release 12.2(13)T. The Cisco 17xx platforms can support only HC DSP firmware images in this release.
12.3(8)T
Documentation of the ds0-group command was divided into the individual ds0-group (E1) and ds0-group (T1) commands.
12.4(2)T1
Support was added for the e&m-lmr signaling type on the Cisco 2691, Cisco 2600XM series, Cisco 2800 series (except Cisco 2801), Cisco 3660, Cisco 3700 series, and Cisco 3800 series.
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows:
NoteThis command does not support the extended echo canceller (EC) feature on the Cisco AS5x00 series.
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 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-startThe 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-startThe 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 mgcpIn 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-startIf a third DS0 group is added, the voice-port is rejected even though the total number of voice channels is fewer than 16.
ds0-group 2 timeslots 17-18 type e&m-immediate-startIn the following example, the signaling type is set to E&M-LMR:
ds0-group 0 timeslots 1-10 type e&m-lmrds0-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 | em-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 { em-delay-dial | em-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 for Cisco AS5300, Cisco AS5350, and Cisco AS5400
ds0-group ds0-group-number timeslots timeslot-list [ service service-type ] [ type 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-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-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-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
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:
typenone
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 a PBX interface. 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:
- e&m-delay-dial --The originating endpoint sends an off-hook signal and then waits for an off-hook signal followed by an on-hook signal from the destination.
- e&m-fgb --E&M Type II Feature Group B.
- e&m-fgd --E&M Type II Feature Group D.
- e&m-immediate-start --E&M immediate start.
- e&m-lmr --E&M Land Mobile Radio (LMR).
- e&m-wink-start --The originating endpoint sends an off-hook signal and waits for a wink-start from the destination.
- ext-sig --The external signaling interface specifies that the signaling traffic comes from an outside source.
- fgd-eana --Feature Group D exchange access North American.
- fgd-emf-- FGD Enhanced MF.
- fgd-os --Feature Group D operator services.
- fxo-ground-start --FXO ground-start signaling.
- fxo-loop-start --FXO loop-start signaling.
- fxs-ground-start --FXS ground-start signaling.
- fxs-loop-start --FXS loop-start signaling.
- none --Null signaling for external call control.
- r1-itu --Line signaling based on international signaling standards. (This signaling type is not supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 platforms.)
- r1-modified --An international signaling standard that is common to channelized T1/E1 networks.
service service -type
(Optional) Specifies the type of service:
- data --Data service.
- fax -- Store-and-forward fax service.
- mgcp --Media Gateway Control Protocol (MGCP) service. Used only with the type none keywords on the Cisco AS5x00 platforms.
- sccp ds0-group (T1)--Simple Gateway Control Protocol (SCCP) service.
- voice --Voice service (for FGD-OS 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 information digits on the Cisco AS5x00 platforms.
Command History
Release
Modification
11.2
This command was introduced for the Cisco AS5300 as the cas-group command.
11.3(1)MA
The command was introduced as the voice-group command for the Cisco MC3810.
12.0(1)T
This command was integrated into Cisco IOS Release 12.0(1)T, and the cas-group command was implemented on the Cisco 3600 series routers.
12.0(5)T
The command was renamed ds0-group on the Cisco AS5300 and Cisco 2600 series and Cisco 3600 series routers. Some keyword modifications were implemented.
12.0(5)XE
This command was implemented on the Cisco 7200 series.
12.0(7)XK
Support for this command was implemented on the Cisco MC3810. When the ds0-group command became available on the Cisco MC3810, the voice-group command was removed and no longer supported. The ext-sig keyword replaced the ext-sig-master and ext-sig-slave keywords that were available with the voice-group command.
12.0(7)XR
The mgcp service type was added.
12.1(2)XH
The e&m-fgd and fgd-eana keywords were added for Feature Group D signaling.
12.1(5)XM
The sgcp keyword was removed.
12.1(3)T
This command was modified for Cisco 7500 series routers. The fgd-os signaling type and the voice service type were added.
12.2(2)XA
This command was implemented on the Cisco AS5300.
12.2
The command was modified to exclude sas keywords. The Single Attachment Station (SAS) CAS options of sas-loop-start and sas-ground-start are not supported as a type of signaling for the DS0 group.
12.2(2)T
This command was integrated into Cisco IOS Release 12.2(2)T and implemented on the Cisco 7200 series.
12.2(4)T
Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on Cisco 1750 and Cisco 1751 routers. Support for other Cisco platforms is not included in this release.
12.2(2)XN
Support for the mgcp keyword was added to Cisco CallManager Version 3.1 for the Cisco 2600 series, Cisco 3600 series, and Cisco VG200.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
12.2(11)T
This command was supported in Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2. This command is supported on the Cisco IAD2420 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5850 in this release.
12.2(13)T
This command was integrated into Cisco IOS Release 12.2(13)T. The Cisco 1750 and Cisco 1751 do not support T1 and E1 voice and data cards in Cisco IOS Release 12.2(13)T. The Cisco 17xx platforms can support only HC DSP firmware images in this release.
12.2(15)T
This command was implemented on the Cisco 2600XM, Cisco 3725, and Cisco 3745.
12.3(4)XD
This command was modified for the Cisco 3725 and Cisco 3745. The e&m-lmr signaling type was added.
12.3(7)T
This command was integrated into Cisco IOS Release 12.3(7)T.
12.3(8)T
Documentation of the ds0-group command was divided into the individual ds0-group(E1) and ds0-group(T1) commands.
12.3(10)
The info-digits-no-strip keyword was added for use on the Cisco AS5x00 platforms.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T. The fgd-emf, ani-pani, and ani keywords were added for the Cisco 2800 and Cisco AS5x00 platforms.
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows:
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:
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 MGCP is configured, you see the mgcp keyword. If you are not using MGCP, you do not see the mgcp keyword.
NoteThis command does not support the extended echo canceller (EC) feature on the Cisco AS5x00 series.
NoteThe 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-startThe 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-startThe 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 service mgcp type noneIn 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-startIf a third DS0 group is added, the voice port is rejected even though the total number of voice channels is fewer than 16.
ds0-group 2 timeslots 17-18 type e&m-immediate-startIn the following example, the signaling type is set to E&M LMR:
ds0-group 0 timeslots 1-10 type e&m-lmrYou 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-stripThe following example enables FGD EMF:
ds0-group 11 timeslots 11 type fgd-emf ani ds0-group 11 timeslots 11 type fgd-emf ani-panids0-num
To add B-channel information in outgoing Session Initiation Protocol (SIP) messages, use the ds0-numcommand in SIP voice service configuration mode. To return to the default setting, use the no form of this command.
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. Information acquired from the Via header can be used during call transfer or to route a call.
dscp media
To specify the resource priority header (RPH) to differentiated services code point (DSCP) mapping, use the dscp media command in voice class configuration mode. To disable the configuration, use the no form of this command.
dscp media { audio | video } { flah-override-override | flash-override | flsh | immediate | priority | routine } { dscp-value | set-af | set-cs | ef | zero }
no dscp media { audio | video } { flah-override-override | flash-override | flsh | immediate | priority | routine } { dscp-value | set-af | set-cs | ef | zero }
Syntax Description
audio
Applies DSCP to audio payload packets.
video
Applies DSCP to video payload packets.
flah-override-override
Applies flash-override-override RPH priority.
flash-override
Applies flash-override RPH priority.
flsh
Applies flash RPH priority.
immediate
Applies immediate RPH priority.
priority
Applies priority RPH priority.
routine
Applies routine RPH priority.
dscp-value
DSCP value. Valid values are from 0 to 63.
set-af
set-cs
ef
Specifies the expedited forwarding bit pattern 101110 as the DSCP value.
zero
Specifies the default bit pattern 000000 as the DSCP value.
Usage Guidelines
You can use the dscp media command to configure RPH to DSCP mapping for audio and video calls.
The following table lists the default values for the dscp media command:
Granular Service Class
Priority or Precedence
DSCP Base10 Value
DSCP Binary Value
Voice
Audio Call
46
101110
Flash
43
101011
Flash Override
41
101001
Flash Override Override
40
101000
Immediate
45
101101
Priority
47
101111
Routine
49
110001
Video
Flash Override
33
100001
Flash
35
100011
Flash Override Override
32
100000
Immediate
37
100101
Priority
39
100111
Routine
51
110011
Video Call
34
100111
dscp-profile
To apply a differentiated services code point (DSCP) profile globally, use the dscp-profile command in voice service SIP configuration mode. To disable the configuration, use the no form of this command.
Usage Guidelines
You can use the dscp-profile command to apply a DSCP profile that is configured using the dscp media command at the global level.
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.
Command History
Release
Modification
12.0(4)XJ
This command was introduced.
12.1(1)T
This command was integrated into Cisco IOS Release 12.1(1)T.
12.2(4)T
This command was implemented on the Cisco 1750.
12.2(8)T
This command was implemented on the Cisco 1751, Cisco 2600 series and Cisco 3600 series, Cisco 3725, and Cisco 3745.
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.
NoteIn 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.
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.
NoteDSNs 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.sample.com> rcpt to: <fax=555-0112@sample.com> NOTIFY=SUCCESS,FAILURE,DELAYThree different states 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 f ailure keywordas part of the dsn command configuration.
This command applies to on-ramp store-and-forward fax functions.
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.
Command Default
Selection of a DSP for signaling channel allocation is based on the internal weighted value assigned to the DSPs.
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.
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.
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.
NoteUse 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)# exitThe 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)# exitThe 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)# exitdspfarm (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.
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 services 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)# dspfarmRelated Commands
Command
Description
dsp services dspfarm
Specifies the NM-HDV or NM-HDV-FARM on which DSP-farm services are to be enabled.
dspfarm transcoder maximum sessions
Specifies the maximum number of transcoding sessions to be supported by a DSP farm.
show dspfarm
Displays summary information about DSP resources.
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.
Command History
Release
Modification
12.1(5)XM
This command was introduced on the Cisco 3660.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T.
12.2(2)XB
This command was implemented on the Cisco 2600 series routers.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T.
12.2(15)T
This command was implemented on the Cisco 2600XM, Cisco 3725, and Cisco 3745.
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 page 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 dspfarmThe following example makes available the DSP resources on an AIM on a modular access router:
voice-card 0 dspfarmRelated Commands
Command
Description
ds0-group
Specifies the DS0 time slots that make up a logical voice port on a T1 or E1 controller, Specifies the signaling type by which the router communicates with the PBX or PSTN, Defines T1or E1 channels for compressed voice calls and the CAS method by which the router connects to the PBX or PSTN.
pri-group
Specifies ISDN PRI on a channelized T1 or E1 controller.
voice-card
Enters voice-card configuration mode.
dspfarm confbridge maximum
To specify the maximum number of concurrent conference sessions for which digital signal processor (DSP) farm resources should be allocated, use the dspfarm confbridge maximumcommand in global configuration mode. To reset to the default, use the no form of this command.
dspfarm confbridge maximum { mixed-mode sessions | sessions } number
no dspfarm confbridge maximum { mixed-mode sessions | sessions } number
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 sessionscommand.
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 intervalcommand in global configuration mode. To reset to the default, use the no form of this command.
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.
dspfarm profile
To enter DSP farm profile configuration mode and define a profile for digital signal processor (DSP) farm services, use the dspfarm profilecommand 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 History
Release
Modification
12.3(8)T
This command was introduced.
12.4(11)XW
The security keyword was added.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
12.4(22)T
Support for IPv6 was added.
15.0(1)M2 15.1(1)T
Support was modified for the Cisco IAD 2430, IAD 2431, IAD 2432, and IAD 2435, and the Cisco VG 202, VG 204, and VG 224 platforms.
Cisco IOS XE Release 3.2S
This command was modified. Support was added to the Cisco ASR 1000 Series Router. The conference, mtp& securitykeywords are not supported on the Cisco ASR 1000 Series Router in this release.
15.1(4)M
This command was modified. The video keyword was added.
Cisco IOS XE Release 3.2S
This command was integrated into Cisco IOS XE Release 3.3S.
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 shutdowncommand 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 identifieruniquely identifies a profile. If the service type and profile identifierare 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.
NoteThe 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 command to be less than 100 percent.
To enable dspfarm profiles for voice services, you must use the dsp services dspfarmcommand under the voice-card submode.
Examples
The following example enables DSP farm services profile 20 for conferencing:
Router(config)# dspfarm profile 20 conferenceNote 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 idThe following example enables DSP farm services profile 1 for transcoding:
Router(config)# dspfarm profile 1 transcodeExamples
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 4Related Commands
Command
Description
dsp service dspfarm
Configures the DSP farm services for a specified voice card.
shutdown (DSP farm profile)
Disables the DSP farm profile.
voice-card
Enters voice card configuration mode
voice-service dsp-reservation
Configures the percentage of DSP resources are reserved for voice services and enables video services to use the remaining DSP resources.
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.
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.
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.
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 sessionscommands.
The maximum number of transcoding sessions depends upon DSP availability in the DSP farm. A single DSP supports four transcoding sessions transmitted 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)# dspfarmRelated Commands
Command
Description
dspfarm (DSP farm)
Enables DSP-farm service.
dspfarm confbridge maximum sessions
Specifies the maximum number of conferencing sessions to be supported by a DSP farm.
dsp services dspfarm
Specifies the NM-HDV or NM-HDV-FARM on which DSP-farm services are to be enabled.
show dspfarm
Displays summary information about DSP resources.
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.
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 enter voice-card configuration mode for a particular voice card, from global configuration mode enter the voice-cardcommand and the slot number for the AIM or network module that you want to add to the pool. See the voice-cardcommand page 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 1/0To 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-extendedRelated Commands
Command
Description
ds0-group
Specifies the DS0 time slots that make up a logical voice port on a T1 or E1 controller.
no shutdown
Disables the interface.
pri-group
Specifies an ISDN PRI on a channelized T1 or E1 controller
show interfaces dspfarm dsp
Displays information about the DSP interface.
voice-card
Enters voice-card configuration mode.
dtmf-interworking
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) or to generate RFC 4733 compliance RTP Named Telephony Event (NTE) packets from Cisco UBE, use the dtmf-interworking command in voice service or dial peer voice configuration mode. To remove the delay interval, use the no form of this command.
Syntax Description
rtp-nte
Enables a delay between the dtmf-digit begin and dtmf-digit end events of RTP NTE packets.
standard
Generates RTP NTE packets that are RFC 4733 compliant.
system
Specifies the default global dual tone multifrequency (DTMF) interworking configuration. This keyword is available only in dial peer voice configuration mode.
Command Default
RFC 2833 packet is sent in a single burst of three dtmf-digit begin events, one duration equaling 50 ms, and three dtmf-digit end events with a duration of 100 ms.
Command Modes
Voice service configuration (config-voi-serv)
Dial peer voice configuration (config-dial-peer)
Usage Guidelines
- dtmf-interworking rtp-nte—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.
- dtmf-interworking standard—When the remote system needs RFC 4733 packets, then use this command to generate RFC 4733 compliance.
- dtmf-interworking system—When this command is configured in dial peer voice configuration mode then the global level dtmf-interworking configuration is applicable. This is the default configuration under the dial peer.
Examples
The following example shows configuration of a delay between the dtmf-digit and events:
Device> enable Device# configure terminal Device(config)# voice service voip Device(config-voi-serv)# dtmf-interworking rtp-nte Device(config-voi-serv)# endThe following example shows the generation of RTP NTE packets that are RFC 4733 compliant:
Device> enable Device# configure terminal Device(config)# voice service voip Device(config-voi-serv)# dtmf-interworking standard Device(config-voi-serv)# enddtmf 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.
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.
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.
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-relayRelated Commands
Command
Description
called-number (dial peer)
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
codec (dial peer)
Specifies the voice coder rate of speech for a VoFR dial peer.
connection
Specifies a connection mode for a voice port.
cptone
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
destination-pattern
Specifies the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
preference
Indicates the preferred order of a dial peer within a rotary hunt group.
session protocol
Establishes a session protocol for calls between the local and remote routers via the packet network.
session target
Specifies a network-specific address for a specified dial peer or destination gatekeeper.
signal-type
Sets the signaling type to be used when connecting to a dial peer.
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-relaycommand 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] [sip-info] [sip-kpml]
no dtmf-relay
Syntax Description
cisco -rtp
Forwards DTMF tones by using Real-Time Transport Protocol (RTP) with a Cisco proprietary payload type.
h245 -alphanumeric
Forwards DTMF tones by using the H.245 "alphanumeric" User Input Indication method. Supports tones from 0 to 9, *, #, and from A to D.
h245 -signal
Forwards DTMF tones by using the H.245 "signal" User Input Indication method. Supports tones from 0 to 9, *, #, and from A to D.
rtp -nte
Forwards DTMF tones by using RTP with the Named Telephone Event (NTE) payload type.
digit-drop
Passes digits out-of-band and drops in-band digits.
Note The digit-dropkeyword is only available when the rtp-nte keyword is configured.
sip-info
Forwards DTMF tones using SIP INFO messages. This keyword is available only if the VoIP dial peer is configured for SIP.
sip-kpml
Forwards DTMF tones using SIP KPML over SIP SUBSCRIBE/NOTIFY messages. This keyword is available only if the VoIP dial peer is configured for SIP.
sip-notify
Forwards DTMF tones using SIP NOTIFY messages. This keyword is available only if the VoIP dial peer is configured for SIP.
Command Default
DTMF tones are disabled and sent in-band. That is, they are left in the audio stream.
Command History
Release
Modification
11.3(2)NA
This command was introduced on the Cisco AS5300.
12.0(2)XH
The cisco-rtp, h245-alphanumeric, and h245-signalkeywords were added.
12.0(5)T
This command was integrated into Cisco IOS Release 12.0(5)T.
12.0(7)XK
This command was first supported for VoIP on the MC3810.
12.1(2)T
Changes made in Cisco IOS Release 12.0(7)XK were integrated into Cisco IOS Release 12.1(2)T.
12.2(8)T
This command was implemented on the Cisco 1751, Cisco 2600 series and Cisco 3600 series, Cisco 3725, and Cisco 3745.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(4)T
This command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 was not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850 platform.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T.
12.2(15)ZJ
The sip-notify keyword was added.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.3(11)T
The digit-drop keyword was added.
15.3(3)M
This command was modified. The sip-info and sip-kpml keywords were added. 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-relaycommand using the rtp-nteand 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.
You can configure dtmf-relay sip-info only if the allow-connections sip to sip command is enabled at the global level.
For SIP, the gateway chooses the format according to the following priority:
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:
- cisco-rtp (highest priority)
- h245-signal
- h245-alphanumeric
- rtp-nte
- 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.
NoteThe 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 keywordis 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-rtpThe 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-signalThe 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-relayThe 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-dropThe 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-nteThe 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-notifyThe following example configures the gateway to send DTMF tones using SIP INFO messages to dial peer 10:
dial-peer voice 10 voip dtmf-relay sip-infodualtone
To enter cp-dualtone configuration mode for specifying a custom call-progress tone, use the dualtone command in custom-cpto ne 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 }
Command History
Release
Modification
12.1(5)XM
This command was introduced on the Cisco 2600 and Cisco 3600 series and on the Cisco MC3810.
12.2(2)T
This command was implemented on the Cisco 1750 router and integrated into Cisco IOS Release 12.2(2)T.
12.4(11)XJ2
The conference keyword was added.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
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 4000The 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 ringbackThe 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 100Related Commands
Command
Description
cadence
Defines the tone on and off durations for a call-progress tone.
conference-join custom-cptone
Defines a custom call-progress tone to indicate joining a conference.
conference-leave custom-cptone
Defines a custom call-progress tone to indicate leaving a conference.
dspfarm profile
Enters DSP farm profile configuration mode and defines a profile for DSP farm services.
frequency
Defines the frequency components for a call-progress tone.
supervisory custom-cptone
Associates a class of custom call-progress tones with a voice port.
voice class custom-cptone
Creates a voice class for defining custom call-progress tones.
