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Contents
R
- radius-server attribute 6
- rai target
- random-contact
- random-request-uri validate
- ras retry
- ras retry lrq
- ras rrq dynamic prefixes
- ras rrq ttl
- ras timeout
- ras timeout decisec
- ras timeout lrq
- rbs-zero
- reason-header override
- recorder profile
- redial
- redirect contact order
- redirect ip2ip (dial peer)
- redirect ip2ip (voice service)
- redirection (SIP)
- refer-ood enable
- referto-passing
- register e164
- registered-caller ring
- registrar
- registrar server
- registration retries
- registration timeout
- registration passthrough
- rel1xx
- remote-party-id
- remote-url
- req-qos
- request
- request peer-header
- request (XML transport)
- reset
- reset timer expires
- resource (voice)
- resource threshold
- resource-pool (mediacard)
- response (voice)
- response (XML application)
- response peer-header
- response size (XML transport)
- response-timeout
- retries (auto-config application)
- retry bye
- retry cancel
- retry comet
- retry interval
- retry invite
- retry keepalive (SIP)
- retry notify
- retry prack
- retry refer
- retry register
- retry rel1xx
- retry response
- retry subscribe
- retry window
- retry-delay
- retry-limit
- ring
- ring cadence
- ring frequency
- ring number
- ringing-timeout
- roaming (dial peer)
- roaming (settlement)
- rrq dynamic-prefixes-accept
- rsvp
- rtcp keepalive
- rtp payload-type
- rtp send-recv
- rtp-ssrc multiplex
- rtsp client session history duration
- rtsp client rtpsetup enable
- rtsp client session history records
- rtsp client timeout connect
- rtsp client timeout message
- rule (ENUM configuration)
- rule (voice translation-rule)
radius-server attribute 6
To provide for the presence of the Service-Type attribute (attribute 6) in RADIUS Access-Accept messages, use the radius-server attribute 6command in global configuration mode. To make the presence of the Service-Type attribute optional in Access-Accept messages, use the no form of this command.
radius-server attribute 6 { mandatory | on-for-login-auth | support-multiple | voice value }
no radius-server attribute 6 { mandatory | on-for-login-auth | support-multiple | voice value }
Syntax Description
Command Default
If this command is not configured, the absence of the Service-Type attribute is ignored, and the authentication or authorization does not fail. The default for the voice keyword is 12.
Usage Guidelines
If this command is configured and the Service-Type attribute is absent in the Access-Accept message packets, the authentication or authorization fails.
The support-multiple keyword allows for multiple instances of the Service-Type attribute to be present in an Access-Accept packet. The default behavior is to disallow multiple instances, which results in an Access-Accept packet containing multiple instances being treated as though an Access-Reject was received.
Examples
The following example shows that the presence of the Service-Type attribute is mandatory in RADIUS Access-Accept messages:
Router(config)# radius-server attribute 6 mandatoryThe following example shows that attribute 6 is to be sent in authentication packets:
Router(config)# radius-server attribute 6 on-for-login-authThe following example shows that multiple Service-Type values are to be supported for each RADIUS profile:
Router(config)# radius-server attribute 6 support-multipleThe following example shows that Service-Type values are to be sent in voice calls:
Router(config)# radius-server attribute 6 voice 1rai target
To configure the Session Initiation Protocol (SIP) Resource Allocation Indication (RAI) mechanism, use the rai target command in SIP UA configuration mode. To disable SIP RAI configuration, use the no form of this command.
rai target target-address resource-group group-index [ transport [ tcp [ tls [ scheme { sip | sips } ] ] | udp ] ]
no rai target target-address
Syntax Description
target-address
IPv4, IPv6, or Domain Name Server (DNS) target address to which the status of the gateway resources are reported. The format of the target address can be one of the following:
- ipv4: ipv4-address
- ipv6: ipv6-address
- dns: domain-name
resource-group
Maps the target address with the resource group index.
group-index
Resource group index. The range is from 1 to 5.
transport
(Optional) Specifies the mechanism to transport the RAI information.
tcp
(Optional) Transports the RAI information through Transmission Control Protocol (TCP).
tls
(Optional) Transports the RAI information through Transport Layer Security (TLS).
scheme
(Optional) Specifies the URL scheme for outgoing messages.
sip
(Optional) Selects SIP URL in outgoing OPTIONS message.
sips
(Optional) Selects Secure SIP (SIPS) URL in outgoing OPTIONS message.
udp
(Optional) Transports the RAI information through Unified Datagram Protocol (UDP).
Usage Guidelines
Use the rai target command to provide the details of SIP along with the index of the resource group that needs to be monitored for reporting over SIP trunk. A maximum of five RAI configurations can be applied for other destination targets or monitoring entities. However, only one RAI configuration is possible for one target address.
Examples
The following example shows how to enable reporting of SIP RAI information over TCP to a target address of example.com:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# rai target dns:example.com resource-group 1Related Commands
Command
Description
debug rai
Enables debugging for Resource Allocation Indication (RAI).
periodic-report interval
Configures periodic reporting parameters for gateway resource entities.
resource (voice)
Configures parameters for monitoring resources, use the resource command in voice-class configuration mode.
show voice class resource-group
Displays the resource group configuration information for a specific resource group or all resource groups.
voice class resource-group
Enters voice-class configuration mode and assigns an identification tag number for a resource group.
random-contact
To populate an outgoing INVITE message with random-contact information (instead of clear-contact information), use the random-contact command in voice service VoIP SIP configuration mode. To disable random-contact information, use the no form of this command.
Usage Guidelines
To populate outbound INVITE messages from the Cisco Unified Border Element with random-contact information instead of clear-contact information, use the random-contact command. This functionality will work only when the Cisco Unified Border Element is configured for Session Initiation Protocol (SIP) registration with random contact using the credentials and registrar commands.
Examples
The following example shows how to populate outbound INVITE messages with random-contact information:
Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# random-contactRelated Commands
Command
Description
credentials (sip ua)
Sends a SIP registration message from a Cisco Unified Border Element in the UP state.
registrar
Enables SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
voice-class sip random-contact
Populates the outgoing INVITE message with random-contact information at the dial-peer level.
random-request-uri validate
To enable the validation of the called number based on the random value generated during the registration of the number, use the random-request-uri validatecommand in voice service VoIP SIP configuration mode. To disable validation, use the no form of this command.
Usage Guidelines
The system generates a random string when registering a new number. An INVITE message with the P-Called-Party-ID value can have the Request-URI set to this random number. To enable the system to identify the called-number from the random number in the Request-URI, use the random-request-uri validate command.
If the P-Called-Party-ID is not set in the INVITE message, the Request URI for that message must contain the called party information (and cannot contain a random number). Therefore validation is performed only on INVITE messages with a P-Called-Party-ID.
Examples
The following example shows how to enable called-number validation at the global configuration level:
Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# random-request-uri validateRelated Commands
Command
Description
credentials (sip ua)
Sends a SIP registration message from a Cisco Unified Border Element in the UP state.
register
Enables SIP gateways to register E.164 numbers on behalf of FXS, EFXS, and SCCP phones with an external SIP proxy or SIP registrar.
voice-class sip random-request-uri validate
Validates the called number based on the random value generated during the registration of the number at the dial-peer configuration level.
ras retry
To configure the H.323 Registration, Admission, and Status (RAS) message retry counters, use the ras retry command in voice service h323 configuration mode. To set the counters to the default values, use the no form of this command.
ras retry { all | arq | brq | drq | grq | rai | rrq } value
no ras retry { all | arq | brq | drq | grq | rai | rrq }
Syntax Description
all
Configures all RAS message counters that do not have explicit values configured individually. If no ras retry all is entered, all values are set to the default except for the individual values that were configured separately.
arq
Configures the admission request (ARQ) message counter.
brq
Configures the bandwidth request (BRQ) message counter.
drq
Configures the disengage request (DRQ) message counter.
grq
Configures the gatekeeper request (GRQ) message counter.
rai
Configures the resource availability indication (RAI) message counter.
rrq
Configures the registration request (RRQ) message counter.
value
Number of times for the gateway to resend messages to the gatekeeper after the timeout period. The timeout period is the period in which a message has not been received by the gateway from the gatekeeper and is configured using the ras timeout command. Valid values are 1 through 30.
Command Default
arq: 2 retries brq: 2 retries drq: 9 retries grq: 2 retries rai: 9 retries rrq: 2 retries
Usage Guidelines
Use this command in conjunction with the ras timeout command. The ras timeout command configures the number of seconds for the gateway to wait before resending a RAS message to a gatekeeper. The ras retry command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gateways and gatekeepers. For example, if you have gatekeepers that are slow to respond to a type of RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
ras retry lrq
To configure the gatekeeper Registration, Admission, and Status (RAS) message retry counters, use the ras retry lrq command in gatekeeper configuration mode. To set the counters to the default values, use the no form of this command.
Syntax Description
lrq
Configures the location request (LRQ) message counter.
value
Number of times for the zone gatekeeper (ZGK) to resend messages to the directory gatekeeper (DGK) after the timeout period. The timeout period is the period in which a message has not been received by the ZKG from the DGK and is configured using the ras timeout lrq command. Valid values are 1 through 30.
Usage Guidelines
Use this command in conjunction with the ras timeout lrq command. The ras timeout lrq command configures the number of seconds for the gateway to wait before resending a RAS message to a gatekeeper. The ras retry lrqcommand configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gateways and gatekeepers. For example, if you have gatekeepers that are slow to respond to a type of RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
ras rrq dynamic prefixes
To enable advertisement of dynamic prefixes in additive registration request (RRQ) RAS messages on the gateway, use the ras rrq dynamic prefixes command in voice service h323 configuration mode. To disable advertisement of dynamic prefixes in additive RRQ messages, use the no form of this command.
Command Default
In Cisco IOS Release 12.2(15)T, the default was set to enabled. In Cisco IOS Release 12.3(3), the default is set to disabled.
Usage Guidelines
In Cisco IOS Release 12.2(15)T, the default for the ras rrq dynamic prefixes command was set to enabled so that the gateway automatically sent dynamic prefixes in additive RRQ messages to the gatekeeper. Beginning in Cisco IOS Release 12.3(3), the default is set to disabled, and you must specify the command to enable the functionality.
ras rrq ttl
To configure the H.323 Registration, Admission, and Status (RAS) registration request (RRQ) time-to-live value, use the ras rrq ttl command in voice service h323 configuration mode. To set the RAS RRQ time-to-live value to the default value, use the no form of this command.
Syntax Description
time-to-live seconds
Number of seconds that the gatekeeper should consider the gateway active. Valid values are 15 through 4000. The time-to-live seconds value must be greater than the margin seconds value.
margin seconds
(Optional) The number of seconds that an RRQ message can be transmitted from the gateway before the time-to-live seconds value advertised to the gatekeeper. Valid values are 1 through 60. The margin time value times two must be less than or equal to the time-to-live seconds value.
Usage Guidelines
Use this command to configure the number of seconds that the gateway should be considered active by the gatekeeper. The gateway transmits this value in the RRQ message to the gatekeeper. The margin time keyword and argument allow the gateway to transmit an early RRQ to the gatekeeper before the time-to-live value advertised to the gatekeeper.
ras timeout
To configure the H.323 Registration, Admission, and Status (RAS) message timeout values, use the ras timeout command in voice service h323 configuration mode. To set the timers to the default values, use the no form of this command.
ras timeout { all | arq | brq | drq | grq | rai | rrq } seconds
no ras timeout { all | arq | brq | drq | grq | rai | rrq }
Syntax Description
all
Configures message timeout values for all RAS messages that do not have explicit values configured individually. If no ras timeout all is entered, all values are set to the default except for the individual values that were configured separately.
arq
Configures the admission request (ARQ) message timer.
brq
Configures the bandwidth request (BRQ) message timer.
drq
Configures the disengage request (DRQ) message timer.
grq
Configures the gatekeeper request (GRQ) message timer.
rai
Configures the resource availability indication (RAI) message timer.
rrq
Configures the registration request (RRQ) message timer.
seconds
Number of seconds for the gateway to wait for a message from the gatekeeper before timing out. Valid values are 1 through 45.
Command Default
arq : 3 secondsbrq: 3 secondsdrq: 3 secondsgrq: 5 secondsrai: 3 secondsrrq: 5 seconds
Usage Guidelines
Use this command in conjunction with the ras retry command. The ras timeout command configures the number of seconds for the gateway to wait before resending a RAS message to a gatekeeper. The ras retry command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gateways and gatekeepers. For example, if you have gatekeepers that are slow to respond to a type of RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
ras timeout decisec
To configure the H.323 Registration, Admission, and Status (RAS) message timeout values in deciseconds, use the ras timeout decisec command in voice service h323 configuration mode. To set the timers to the default values, use the no form of this command.
ras timeout { all | arq | brq | drq | grq | rai | rrq } decisec decisecond
no ras timeout { all | arq | brq | drq | grq | rai | rrq } decisec
Syntax Description
all
Configures message timeout values for all RAS messages that do not have explicit values configured individually. If no ras timeout all is entered, all values are set to the default except for the individual values that were configured separately.
arq
Configures the admission request (ARQ) message timer. Default: 3.
brq
Configures the bandwidth request (BRQ) message timer. Default: 3.
drq
Configures the disengage request (DRQ) message timer.Default: 3.
grq
Configures the gatekeeper request (GRQ) message timer. Default: 5.
rai
Configures the resource availability indication (RAI) message timer. Default: 3.
rrq
Configures the registration request (RRQ) message timer. Default: 5.
decisecond
Number of deciseconds for the gateway to wait for a message from the gatekeeper before timing out. Valid values are 1 through 45.
Usage Guidelines
Use this command in conjunction with the ras retry command. The ras timeout decisec command configures the number of deciseconds for the gateway to wait before resending a RAS message to a gatekeeper. The ras retry command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gateways and gatekeepers. For example, if you have gatekeepers that are slow to respond to a type of RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
ras timeout lrq
To configure the Gatekeeper Registration, Admission, and Status (RAS) message timeout values, use the ras timeout lrq command in gatekeeper configuration mode. To set the timers to the default values, use the no form of this command.
Usage Guidelines
Use this command in conjunction with the ras retry lrq command. The ras timeout lrq command configures the number of seconds for the zone gatekeeper (ZGK) to wait before resending a RAS message to a directory gatekeeper (DGK). The ras retry lrq command configures the number of times to resend the RAS message after the timeout period expires. The default values for timeouts and retries are acceptable in most networks. You can use these commands if you are experiencing problems in RAS message transmission between gatekeepers. For example, if you have gatekeepers that are slow to respond to a LRQ RAS request, increasing the timeout value and the number of retries increases the call success rate, preventing lost billing information and unnecessary switchover to an alternate gatekeeper.
rbs-zero
To enable 1AESS switch support for T1 lines on the primary serial interface of an access server, use the rbs-zerocommand in serial interface configuration mode. To disable IAESS switch support, use the no form of this command.
Command History
Release
Modification
12.2(2)XA
This command was introduced.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and 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 supports the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5800.
Usage Guidelines
Use this command to configure the primary serial interface of an access server connected to T1 lines to support 1AESS switches for dial-in and dial-out calls. Modem calls of 56K or a lower rate are accepted; 64K calls are rejected.
In IAESS mode, the following occurs:
- Modem calls are accepted and digital calls are rejected.
- The ABCD bit of the 8 bits in the incoming calls is ignored. The ABCD bit of the 8 bits in the outgoing modem calls is set to 0.
In non-1AESS mode, modem and digital calls are accepted.
Examples
The following example enables 1AESS switching support on T1 channel 0:
Router(config)# controller t1 1/0 Router(config-controller)# framing esf Router(config-controller)# linecode b8zs Router(config-controller)# pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 1 Router(config)# interface serial 1/0:23 Router(config-if)# no ip address Router(config-if)# isdn switch-type primary-ni Router(config-if)# rbs-zero nfas-int 0Related Commands
Command
Description
interface serial
Enters serial interface configuration mode.
isdn switch -type
Sets the switch type.
pri -group timeslots
Configures the PRI trunk for a designated operation.
show controllers t1
Displays information about the T1 links and the hardware and software driver information for the T1 controller.
show isdn nfas group
Displays all the members of a specified NFAS group or all NFAS groups.
reason-header override
To enable cause code passing from one SIP leg to another, use the reason-header overridecommand in SIP UA configuration mode. To disable reason-header override, use the no form of this command.
Usage Guidelines
In an SIP-to-SIP configuration the reason-header overridecommand must be configured to ensure cause code passing from the incoming SIP leg to the outgoing SIP leg.
recorder profile
To configure a media profile recorder, use the recorder profile command in media class configuration mode. To disable the configuration, use the no form of this command.
Usage Guidelines
Use the recorder profile command to associate a recorder profile with a media class. The configured recorder profile specifies the recorder profile that is used by the media class. You can configure any number of recorder profiles.
redial
To define speed-dial code for a Feature Speed-dial (FSD) to redial the last number dialed, use the redial command in STC application feature speed-dial configuration mode. To return the code to its default, use the no form of this command.
Syntax Description
keypad-character
Character string that can be dialed on a telephone keypad (0-9, *, #). Default: #.
Before Cisco IOS Release 12.4(20)YA, this is a single character. In Cisco IOS Release 12.5(20)YA and later releases, the string can be any of the following:
- A single character (0-9, *, #)
- Two digits (00-99)
- Two to four characters (0-9, *, #) and the leading or ending character must be an asterisk (*) or number sign (#)
Usage Guidelines
This command changes the value of the speed-dial code for Redial from the default (#) to the specified value.
In Cisco IOS Release 12.4(20)YA and later releases, if the length of the keypad-character argument is at least two characters and the leading or ending character of the string is an asterisk (*) or a number sign (#), phone users are not required to dial a prefix to access this speed dial. Typically, phone users dial a Feature Speed-dial (FSD) consisting of a prefix plus a speed-dial code, for example *#. If the feature code is 78#, the phone user dials only 78#, without the FSD prefix, to access the corresponding feature.
In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that is already being used for a feature access code (FAC) or another FSD, you receive a message. If you configure a duplicate code, the system implements the first matching feature in the order of precedence shown in the output of the show stcapp feature codes command.
In Cisco IOS Release 12.4(20)YA and later releases, if you attempt to configure this command with a value that precludes or is precluded by a feature code for a FAC or another FSD, you receive a message. If you configure this command with a value that precludes or is precluded by another code, 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. You must configure a new value for the precluded code in order to enable access to that feature.
To display a list of all FACs and FSDs, use the show stcapp feature codes command.
Examples
The following example shows how to change the value of the speed-dial code for Redial from the default (#). In this configuration, a phone user must press ** on the keypad to redial the number that was most recently dialed on this line, regardless of what value is configured for the FSD prefix.
Router(config)# stcapp feature speed-dial Router(config-stcapp-fsd)# redial ** Router(config-stcapp-fsd)# exitRelated Commands
Command
Description
digit
Designates the number of digits for feature speed-dial codes (FSDs).
prefix (stcapp-fsd)
Defines the prefix for feature speed-dials (FSDs).
show stcapp feature codes
Displays all feature access codes (FACs) and feature access codes (FSDs) that are available for the STC application.
speed dial
Designates a range of speed-dial codes for the STC application.
stcapp feature speed-dial
Enables feature speed-dials (FSDs) in STC application and enters STC application feature speed-dial configuration mode for changing values of the prefix and speed-dial codes from the default.
redirect contact order
To set the order of contacts in the 300 Multiple Choice message, use the redirect contact order command in SIP configuration mode. To reset the order of contacts to the default, use the no form of this command.
Usage Guidelines
This command applies when a 300 Multiple Choice message is sent by a SIP gateway indicating that a call has been redirected and that there are multiple routes to the destination.
Enter SIP configuration mode after entering voice service VoIP configuration mode as shown in the following example.
redirect ip2ip (dial peer)
To redirect SIP phone calls to SIP phone calls on a specific VoIP dial peer using the Cisco IOS Voice Gateway, use the redirect ip2ip command in dial peer configuration mode. To disable redirection, use the no form of this command.
Usage Guidelines
The redirect ip2ipcommand must be configured on the inbound dial peer of the gateway. This command enables, on a per dial peer basis, IP-to-IP call redirection for the gateway.
To enable global IP-to-IP call redirection for all VoIP dial peers, use voice service configuration mode. To specify IP-to-IP call redirection for a specific VoIP dial peer, configure the dial peer in dial-peer configuration mode.
NoteWhen IP-to-IP redirection is configured in dial-peer configuration mode, the configuration for the specific dial peer is activated only if the dial peer is an inbound dial peer. To enable IP-to-IP redirection globally, use redirect ip2ip (voice service)command.
redirect ip2ip (voice service)
To redirect SIP phone calls to SIP phone calls globally on a gateway using the Cisco IOS Voice Gateway, use the redirect ip2ipcommand in voice service configuration mode. To disable redirection, use the no form of this command.
Usage Guidelines
Use this command to enable IP-to-IP call redirection globally on a gateway. Use the redirect ip2ip(dial-peer) command to configure IP-to-IP redirection on a specific inbound dial peer.
redirection (SIP)
To enable the handling of 3xx redirect messages, use the redirection command in SIP UA configuration mode. To disable the handling of 3xx redirect messages, use the no form of this command.
Usage Guidelines
The redirection command applies to all Session Initiation Protocol (SIP) VoIP dial peers configured on the gateway.
The default mode of SIP gateways is to process incoming 3xx redirect messages according to RFC 2543. However if redirect handling is disabled with the no redirectioncommand, the gateway treats the incoming 3xx responses as 4xx error class responses. To reset the default processing of 3xx messages, use the redirection command.
refer-ood enable
To enable out-of-dialog refer (OOD-R) processing, use the refer-ood enable command in SIP user-agent configuration mode. To disable OOD-R, use the no form of this command.
Usage Guidelines
Out of dialog Refer allows applications to establish calls using the SIP gateway or Cisco Unified CME. The application sets up the call and the user does not dial out from their own phone.
referto-passing
To disable dial peer lookup and modification of the Refer-To header when the Cisco Unified Border Element (UBE) passes across a REFER message during a call transfer, use the referto-passing command in voice service voip SIP configuration mode. To enable dial peer lookup and the Refer-To header modification, use the no form of this command.
Command Default
Dial peer lookup is performed. The Refer-To header is modified to include the address of the Cisco UBE if address hiding is enabled or to include the address of the call target if a dial peer match is found.
Usage Guidelines
By default, while passing across the REFER message, the Cisco UBE replaces the host portion of the Refer-To header with the address of the Cisco UBE if the address-hiding command is enabled or with the address of the call target if a dial peer match is found. You can use the referto-passing command to disable the Cisco UBE from overwriting the Refer-To header even if address hiding is enabled. This command also disables dial peer lookup when the Cisco UBE passes across the REFER message.
Examples
The following example shows how to enable REFER message pass-through on the Cisco UBE and disable the modification of the Refer-To header:
Router(config)# voice service voip Router(conf-voi-serv)# supplementary-service sip refer Router(conf-voi-serv)# sip Router(conf-serv-sip)# referto-passingregister e164
To configure a gateway to register or deregister a fully-qualified dial-peer E.164 address with a gatekeeper, use the register e164command in dial peer configuration mode. To deregister the E.164 address, use the no form of this command.
Command History
Release
Modification
12.0(5)T
This command was introduced.
12.1(5)XM2
The 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 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400, and the Cisco AS5850 in this release.
Usage Guidelines
Use this command to register the E.164 address of an analog telephone line attached to a foreign exchange station (FXS) port on a router. The gateway automatically registers fully qualified E.164 addresses. Use the no register e164command to deregister an address. Use the register e164command to register a deregistered address.
Before you automatically or manually register an E.164 address with a gatekeeper, you must create a dial peer (using the dial-peer command), assign an FXS port to the peer (using the port command), and assign an E.164 address using the destination-pattern command. The E.164 address must be a fully qualified address. For example, +5550112, 5550112, and 4085550112 are fully qualified addresses; 408555.... is not. E.164 addresses are registered only for active interfaces, which are those that are not shut down. If an FXS port or its interface is shut down, the corresponding E.164 address is deregistered.
Tip
You can use the show gateway command to find out whether the gateway is connected to a gatekeeper and whether a fully qualified E.164 address is assigned to the gateway. Use the zone-prefix command to define prefix patterns on the gatekeeper, such as 408555...., that apply to one or more gateways.
Examples
The following command sequence places the gateway in dial peer configuration mode, assigns an E.164 address to the interface, and registers that address with the gatekeeper.
gateway1(config)# dial-peer voice 111 pots gateway1(config-dial-peer)# port 1/0/0 gateway1(config-dial-peer)# destination-pattern 5550112 gateway1(config-dial-peer)# register e164The following commands deregister an address with the gatekeeper.
gateway1(config)# dial-peer voice 111 pots gateway1(config-dial-peer)# no register e164The following example shows that you must have a connection to a gatekeeper and must define a unique E.164 address before you can register an address.
gateway1(config)# dial-peer voice 222 pots gateway1(config-dial-peer)# port 1/0/0 gateway1(config-dial-peer)# destination 919555.... gateway1(config-dial-peer)# register e164 ERROR-register-e164:Dial-peer destination-pattern is not a full E.164 number gateway1(config-dial-peer)# no gateway gateway1(config-dial-peer)# dial-peer voice 111 pots gateway1(config-dial-peer)# register e164 ERROR-register-e164:No gatekeeperRelated Commands
Command
Description
destination -pattern
Specifies either the prefix or the full E.164 telephone number (depending on your dial plan) to be used for a dial peer.
dial -peer (voice)
Enters dial peer configuration mode and specifies the method of voice encapsulation.
port (dial peer)
Associates a dial peer with a specific voice port.
show gateway
Displays the current gateway status.
zone prefix
Adds a prefix to the gatekeeper zone list.
registered-caller ring
To configure the Nariwake service registered caller ring cadence, use the registered-caller ring command in dial peer configuration mode.
Usage Guidelines
If your ISDN line is provisioned for the I Number or dial-in services, you must also configure a dial peer by using the destination-pattern not-provided command. Either port 1 or port 2 can be configured under this dial peer. The router then forwards the incoming call to voice port 1. (See the "Examples" section below.
If more than one dial peer is configured with the destination-pattern not-provided command, the router uses the first configured dial peer for the incoming calls. To display the Nariwake ring cadence setting, use the show run command.
registrar
To enable Session Initiation Protocol (SIP) gateways to register E.164 numbers on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and Skinny Client Control Protocol (SCCP) phones with an external SIP proxy or SIP registrar, use the registrar command in SIP UA configuration mode. To disable registration of E.164 numbers, use the no form of this command.
registrar { dhcp | [registrar-index] registrar-server-address [ : port ] } [ auth-realm realm ] [ expires seconds ] [random-contact] [ refresh-ratio ratio-percentage ] [ scheme { sip | sips } ] [tcp] [type] [secondary]
no registrar [ registrar-index | secondary ]
Syntax Description
dhcp
(Optional) Specifies that the domain name of the primary registrar server is retrieved from a DHCP server (cannot be used to configure secondary or multiple registrars).
registrar-index
(Optional) A specific registrar to be configured, allowing configuration of multiple registrars (maximum of six). Range is 1 to 6.
registrar-server-address
The SIP registrar server address to be used for endpoint registration. This value can be entered in one of three formats:
- dns: address --the Domain Name System (DNS) address of the primary SIP registrar server (the dns: delimiter must be included as the first four characters).
- ipv4: address --the IP address of the SIP registrar server (the ipv4: delimiter must be included as the first five characters).
- ipv6:[ address ]--the IPv6 address of the SIP registrar server (the ipv6: delimiter must be included as the first five characters and the address itself must include opening and closing square brackets).
: port ]
(Optional) The SIP port number (the colon delimiter is required).
auth-realm
(Optional) Specifies the realm for preloaded authorization.
realm
The realm name.
expires seconds
(Optional) Specifies the default registration time, in seconds. Range is 60 to 65535. Default is 3600.
random-contact
(Optional) Specifies the Random String Contact header used to identify the registration session.
refresh-ratio ratio-percentage
(Optional) Specifies the registration refresh ratio, in percentage. Range is 1 to 100. Default is 80.
scheme {sip | sips}
(Optional) Specifies the URL scheme. The options are SIP (sip) or secure SIP (sips), depending on your software installation. The default is sip.
tcp
(Optional) Specifies TCP. If not specified, the default is User Datagram Protocol UDP.
type
(Optional) The registration type.
Note The type argument cannot be used with the dhcp option.
secondary
(Optional) Specifies a secondary SIP registrar for redundancy if the primary registrar fails. This option is not valid if specifying DHCP or if configuring multiple registrars.
Note You cannot configure any other optional settings once you enter the secondary keyword--specify all other settings first.
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.4(6)T
This command was modified. The tls keyword and the scheme keyword with the string argument were added.
12.4(22)T
This command was modified. Support for IPv6 addresses was added.
12.4(22)YB
This command was modified. The dhcp, random-contact and refresh-ratio keywords were added. Additionally, the aor-domain keyword and the tls option for the tcp keyword were removed.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
15.0(1)XA
This command was modified. The registrar-index argument for support of multiple registrars on SIP trunks was added.
15.1(1)T
This command was integrated into Cisco IOS Release 15.1(1)T.
15.1(2)T
This command was modified. The auth-realm keyword was added.
Usage Guidelines
Use the registrar dhcp or registrar registrar-server-address command to enable the gateway to register E.164 telephone numbers with primary and secondary external SIP registrars. In Cisco IOS Release 15.0(1)XA and later releases, endpoints on Cisco IOS SIP time-division multiplexing (TDM) gateways, Cisco Unified Border Elements (Cisco UBEs), and Cisco Unified Communications Manager Express (Cisco Unified CME) can be registered to multiple registrars using the registrar registrar-index command.
By default, Cisco IOS SIP gateways do not generate SIP register messages.
NoteWhen entering an IPv6 address, you must include square brackets around the address value.
Examples
The following example shows how to configure registration with a primary and secondary registrar:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# retry invite 3 Router(config-sip-ua)# retry register 3 Router(config-sip-ua)# timers register 150 Router(config-sip-ua)# registrar ipv4:209.165.201.1 expires 14400 secondaryThe following example shows how to configure a device to register with the SIP server address received from the DHCP server. The dhcp keyword is available only for configuration by the primary registrar and cannot be used if configuring multiple registrars.
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# registrar dhcp expires 14400The following example shows how to configure a primary registrar using an IP address with TCP:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# retry invite 3 Router(config-sip-ua)# retry register 3 Router(config-sip-ua)# timers register 150 Router(config-sip-ua)# registrar ipv4:209.165.201.3 tcpThe following example shows how to configure a URL scheme with SIP security:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# retry invite 3 Router(config-sip-ua)# retry register 3 Router(config-sip-ua)# timers register 150 Router(config-sip-ua)# registrar ipv4:209.165.201.7 scheme sipsThe following example shows how to configure a secondary registrar using an IPv6 address:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# registrar ipv6:[3FFE:501:FFFF:5:20F:F7FF:FE0B:2972] expires 14400 secondaryThe following example shows how to configure all POTS endpoints to two registrars using DNS addresses:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# registrar 1 dns:example1.com expires 180 Router(config-sip-ua)# registrar 2 dns:example2.com expires 360The following example shows how to configure the realm for preloaded authorization using the registrar server address:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# registrar 2 192.168.140.3:8080 auth-realm example.com expires 180Related Commands
Command
Description
authentication (dial peer)
Enables SIP digest authentication on an individual dial peer.
authentication (SIP UA)
Enables SIP digest authentication.
credentials (SIP UA)
Configures a Cisco UBE to send a SIP registration message when in the UP state.
localhost
Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.
retry register
Sets the total number of SIP register messages to send.
show sip-ua register status
Displays the status of E.164 numbers that a SIP gateway has registered with an external primary or secondary SIP registrar.
timers register
Sets how long the SIP UA waits before sending register requests.
voice-class sip localhost
Configures settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages on an individual dial peer, overriding the global setting.
registrar server
To enable the local Session Initiation Protocol (SIP) registrar, use the registrar server command in service SIP configuration mode. To disable the configuration, use the no form of this command.
Syntax Description
expires
(Optional) Configures the registration expiry time.
max value
(Optional) Configures the maximum registration expiry time, in seconds. The range is from 120 to 86400. The default is 3600.
min value
(Optional) Configures the minimum registration expiry time, in seconds. The range is from 60 to 3600. The default is 60.
Usage Guidelines
You must enable the local SIP registrar by using the registrar server command before configuring the SIP registration on Cisco Unified Border Element (UBE).
registration retries
To set the number of times that Skinny Client Control Protocol (SCCP) tries to register with a Cisco Unified CallManager, use the registration retriescommand in SCCP Cisco CallManager configuration mode. To reset this number to the default value, use the no form of this command.
Usage Guidelines
Use this command to control the number of registration retries before SCCP confirms that it cannot register with the Cisco Unified CallManager. When SCCP confirms that it cannot register to the current Cisco Unified CallManager (if the number of registration requests sent without an Ack reaches the registration retries value), SCCP tries to register with the next Cisco Unified CallManager.
NoteThe optimum setting for this command depends on the platform and your individual network characteristics. Adjust the registration retry attempts to meet your needs.
registration timeout
To set the length of time between registration messages sent from Skinny Client Control Protocol (SCCP) to the Cisco Unified CallManager, use the registration timeoutcommand in SCCP Cisco CallManager configuration mode. To reset the length of time to the default value, use the no form of this command.
Usage Guidelines
Whenever SCCP sends the registration message to the Cisco Unified CallManager, it initiates this timer. Once the timeout occurs, it sends the next registration message unless the number of messages without an Ack reaches the number set by the registration retriescommand. Use this command to set the Cisco Unified CallManager registration timeout parameter value.
NoteThe optimum setting for this command depends on the platform and your individual network characteristics. Adjust the registration timeout value to meet your needs.
registration passthrough
To configure the Session Initiation Protocol (SIP) registration pass-through options, use the registration passthrough command in service SIP configuration mode. To disable the configuration, use the no form of this command.
registration passthrough [static] [ rate-limit [ expires value ] [ fail-count value ] ] [ registrar-index [index] ]
no registration passthrough
Syntax Description
static
(Optional) Configures Cisco Unified Border Element (UBE) to use static registrar details for SIP registration. Cisco UBE works in point-to-point mode when the static keyword is used.
rate-limit
(Optional) Configures SIP registration pass-through rate limit options.
expires value
(Optional) Sets the expiry value for rate limiting, in seconds. The range is from 60 to 65535. The default value is 3600.
fail-count value
(Optional) Sets the fail count value for rate limiting. The range is from 2 to 20. The default value is 0.
registrar-index
(Optional) Configures the registrar index that is to be used for registration pass-through.
index
(Optional) Registration index value. The range is from 1 to 6.
Usage Guidelines
You can use the registration passthrough command to configure the following SIP pass-through functionalities:
- Back-to-back registration facility to register phones for call routing.
- Options to configure the rate-limiting values, such as the expiry time, fail-count, and a list of registrars to be used for registration.
rel1xx
To enable all Session Initiation Protocol (SIP) provisional responses (other than 100 Trying) to be sent reliably to the remote SIP endpoint, use the rel1xxcommand in SIP configuration mode. To reset to the default, use the no form of this command.
Syntax Description
supported value
Supports reliable provisional responses. The value argument may have any value, as long as both the user-agent client (UAC) and user-agent server (UAS) configure it the same. This keyword, with value of 100rel, is the default.
require value
Requires reliable provisional responses. The value argument may have any value, as long as both the UAC and UAS configure it the same.
disable
Disables the use of reliable provisional responses.
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco 7200 series. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(11)T
This command was supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Usage Guidelines
The use of resource reservation with SIP requires that the reliable provisional feature for SIP be enabled either at the VoIP dial-peer level or globally on the router.
There are two ways to configure reliable provisional responses:
- Dial-peer configuration mode. You can configure reliable provisional responses for the specific dial peer only by using the voice-class sip rel1xxcommand.
- SIP configuration mode. You can configure reliable provisional responses globally by using the rel1xxcommand.
The voice-class sip rel1xx command in dial-peer configuration mode takes precedence over the rel1xxcommand in global configuration mode with one exception: If the voice-class sip rel1xx command is used with the systemkeyword, the gateway uses what was configured under the rel1xx command in global configuration mode.
Enter SIP configuration mode from voice-service VoIP configuration mode as shown in the following example.
remote-party-id
To enable translation of the SIP header Remote-Party-ID, use the remote-party-id command in SIP UA configuration mode. To disable Remote-Party-ID translation, use the no form of this command.
Usage Guidelines
When the remote-party-id command is enabled, one of the following calling information treatments occurs:
- If a Remote-Party-ID header is present in the incoming INVITE message, the calling name and number extracted from the Remote-Party-ID header are sent as the calling name and number in the outgoing Setup message. This is the default behavior. Use the remote-party-id command to enable this option.
- When no Remote-Party-ID header is available, no translation occurs so the calling name and number are extracted from the From header and are sent as the calling name and number in the outgoing Setup message. This treatment also occurs when the feature is disabled.
Examples
The following example shows the Remote-Party-ID translation being enabled:
Router(config-sip-ua)# remote-party-idremote-url
To configure the url the application that wil be used by the service provider, use the remote-url command. The provider will use this url to authenticate and commnunicate with the application. To delete the configured url, use the no form of this command.
Usage Guidelines
Use this command to configure the remote URL (application) that the service provider uses in messages.
Examples
The following example configures the remote url that the the xcc service provider will use in messages.
Router(config)# uc wsapi Router(config-uc-wsapi)# provider xcc Router(config-uc-wsapi-xcc)# no shutdown Router(config-uc-wsapi-xcc)# remote-url 1 http://209.133.85.47:8090/my_route_controlreq-qos
To specify the desired quality of service to be used in reaching a specified dial peer, use the req-qos command in dial peer configuration mode. To restore the default value for this command, use the no form of this command.
req-qos { best-effort | controlled-load | guaranteed-delay } [ { audio bandwidth | video bandwidth } default | max bandwidth-value ]
no req-qos
Syntax Description
best-effort
Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation.
controlled-load
Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to assure that preferential service is received even when the bandwidth is overloaded.
guaranteed-delay
Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.
audio bandwidth
(Optional) Specifies amount of bandwidth to be requested for audio streams.
default
Sets the default bandwidth to be requested for audio or video streams.
- Audio streams--Range is 1 to 64 kbps; default value is 64 kbps.
- Video streams--Range is 1 to 5000 kbps; default value is no maximum
max bandwidth-value
Sets the maximum bandwidth to be requested for audio streams. Range is 1 to 64 kbps; default value is no maximum.
video bandwidth
(Optional) Specifies the amount of bandwidth to be requested for video streams.
default bandwidth-value
Sets the default bandwidth to be requested for video streams. Range is 1 to 5000 kbps; default value is 384 kbps.
max bandwidth-value
(Optional) Sets the maximum bandwidth to be requested for video streams. .
Usage Guidelines
Use the req-qoscommand to request a specific quality of service to be used in reaching a dial peer. Like acc-qos, when you issue this command, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.
This command is applicable only to VoIP dial peers.
Examples
The following example configures guaranteed-delay as the requested quality of service to a dial peer:
dial-peer voice 10 voip req-qos guaranteed-delayThe following example configures guaranteed-delayandrequests a default bandwidth level of 768 kbps for video streams:
dial-peer voice 20 voip req-qos guaranteed-delay video bandwidth default 768request
To use SIP profiles to add, copy, modify, or remove Session Initiation Protocol (SIP) or Session Description Protocol (SDP) header value in a SIP request message, use the requestcommand in voice class configuration mode. To disable the configuration, use the no form of this command.
request method { sdp-header | sip-header } header-name { add | copy | modify | remove } string
no request method { sdp-header | sip-header } header-name { add | copy | modify | remove } string
Syntax Description
method
Type of message to be added, modified, or removed.
It can be one of the following values:
- ack --SIP acknowledgment message.
- any --Any SIP message.
- bye --SIP BYE message.
- cancel --SIP CANCEL message.
- comet --SIP COMET message.
- info --SIP INFO message.
- invite --The first SIP INVITE message.
- notify --SIP NOTIFY message.
- options --SIP OPTIONS message.
- prack --SIP PRACK message.
- publish --SIP PUBLISH message.
- refer --SIP REFER message.
- register --SIP REGISTER message.
- reinvite --SIP REINVITE message.
- subscribe --SIP SUBSCRIBE message.
- update --SIP UPDATE message.
sdp-header
Specifies an SDP header.
sip-header
Specifies a SIP header.
header-name
SDP or SIP header name.
add
Adds a header.
copy
Copies a header.
modify
Modifies a header.
remove
Removes a header.
string
String to be added, copied, modified, or removed as a header.
Note If you use the copy keyword, you must provide a matching pattern followed by the variable name for the string argument.
Command Default
SIP profiles are not modified to add, copy, modify, or remove SIP or SDP header values.
Usage Guidelines
If there are interoperability issues with Cisco UBE, the Cisco UBE will not work with the default SIP signaling. Hence, you must modify the SIP profiles to add, copy, modify, or remove SIP or SDP header values, and therefore enable Cisco UBE to work with SIP signaling.
Use the requestcommand to modify SIP profiles for a request message. You can add, copy, modify, or remove SIP or SDP header values in an outgoing SIP request message.
request peer-header
To use SIP profiles to copy a peer header from an outgoing Session Initiation Protocol (SIP) request message, use the request peer-header command in voice class configuration mode. To disable the configuration, use the no form of this command.
request method peer-header sip { sip-req-uri | header-name } copy pattern variable
no request method peer-header sip { sip-req-uri | header-name } copy pattern variable
Syntax Description
method
Type of message to be copied.
You can specify any of the following values:
- ack --SIP acknowledgment message.
- any --SIP message.
- bye --SIP BYE message.
- cancel --SIP CANCEL message.
- comet --SIP COMET message.
- info --SIP INFO message.
- invite --First SIP INVITE message.
- notify --Specifies SIP NOTIFY message.
- options --SIP OPTIONS message.
- prack --SIP PRACK message.
- publish --SIP PUBLISH message.
- refer --SIP REFER message.
- register --SIP REGISTER message.
- reinvite --SIP REINVITE message.
- subscribe --SIP SUBSCRIBE message.
- update --SIP UPDATE message.
sip
Specifies that the SIP header must be copied from the peer call leg.
sip-req-uri
Specifies the SIP request Uniform Resource Identifier (URI) to be copied from the peer call leg.
header-name
Header name from which the values must be copied.
copy
Copies a header.
pattern
Match pattern.
variable
Variable to which the pattern value must be copied. The range is from u01 to u99.
Command Default
No SIP profiles are modified to copy a peer header in an outgoing SIP request message.
Usage Guidelines
If there are interoperability issues with Cisco UBE, then the Cisco UBE will not be able to work with the default SIP signaling. Hence, you must modify the SIP profiles to add, copy, modify, or remove SIP or SDP header values, and therefore enable Cisco UBE to work with SIP signaling.
Configure the request peer-headercommand to use SIP profiles to copy a peer header from an outgoing SIP request message.
request (XML transport)
To set the XML transport mode request handling parameters, use the request command in XML transport configuration mode. To disable the XML transport request parameter setting, use the no form of this command
Syntax Description
outstanding
Maximum number of outstanding requests.
number
The valid range for the number of outstanding requests is from 1 to 10. The default is 1.
timeout
Response timeout at the transport level.
seconds
Specifies the number of seconds a request is active before it times out. Valid rangeis from 0 to 60 seconds. The default value is 0 (no timeout).
Usage Guidelines
Use this command to set the request timeout. A value of 0 seconds specifies no timeout. This timeout applies to the request being processed and not outstanding requests as described below. The specified timeout limits the amount of time between the request being dequeued by the application and the completion of the processing of that request.
Use this command to specify the number of outstanding requests allowed per application for the specified transport mode. The outstanding requests are those requests that are queued at the application for processing but have not yet been processed.
reset
reset timer expires
To globally configure Cisco Unified Communications Manager Express (Cisco Unified CME), a Cisco IOS voice gateway, or a Cisco Unified Border Element (Cisco UBE) to reset the expires timer upon receipt of a Session Initiation Protocol (SIP) 183 Session In Progress message, use the reset timer expires command in voice service SIP configuration mode. To globally disable resetting of the expires timer upon receipt of SIP 183 messages, use the no form of this command.
Command Default
The expires timer is not reset after receipt of SIP 183 Session In Progress messages and a session or call that is not connected within the default expiration time (three minutes) is dropped.
Usage Guidelines
In some scenarios, early media cut-through calls (such as emergency calls) rely on SIP 183 with session description protocol (SDP) Session In Progress messages to keep the session or call alive until receiving a FINAL SIP 200 OK message, which indicates that the call is connected. In these scenarios, the call can time out and be dropped if it does not get connected within the default expiration time (three minutes).
NoteThe expires timer default is three minutes. However, you can configure the expiration time to a maximum of 30 minutes using the timers expires command in SIP user agent (UA) configuration mode.
To prevent early media cut-through calls from being dropped because they reach the expires timer limit, use the reset timer expires command in voice service SIP configuration mode to globally enable all dial peers on Cisco Unified CME, Cisco IOS voice gateways, or Cisco UBEs to reset the expires timer upon receipt of any SIP 183 message.
To configure the reset timer expiration setting for an individual dial peer, use the voice-class sip reset timer expires command in dial peer voice configuration mode. To disable the expires timer reset on receipt of SIP 183 messages function, use the no reset timer expires command in voice service SIP configuration mode.
Examples
The following example shows how to globally configure all dial peers on Cisco Unified CME, a Cisco IOS voice gateway, or a Cisco UBE to reset the expires timer each time a SIP 183 message is received:
Router> enable Router# configure terminal Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# reset timer expires 183Related Commands
Command
Description
timers expires
Specifies how long a SIP INVITE request remains valid before it times out if no appropriate response is received for keeping the session alive.
voice-class sip reset timer expires
Configures an individual dial peer on Cisco Unified CME, a Cisco IOS voice gateway, or a Cisco UBE to reset the expires timer upon receipt of a SIP 183 message.
resource (voice)
To configure parameters for monitoring resources, use the resource command in voice-class configuration mode. To disable the configuration for monitoring resources, use the no form of this command.
resource { cpu { 1-min-avg | 5-sec-avg } | ds0 | dsp | mem { io-mem | proc-mem | total-mem } } [ threshold high threshold-value low threshold-value ]
no resource { cpu | ds0 | dsp | mem }
Syntax Description
cpu
Reports the CPU utilization information.
1-min-avg
Collects the CPU data for an average of one minute.
5-sec-avg
Collects the CPU data for an average of five seconds.
ds0
Reports utilization information for the DS0 port.
dsp
Reports utilization information for the digital signal processor (DSP) channel.
mem
Reports the memory utilization information.
io-mem
Reports the input/output memory utilization information.
proc-mem
Reports the process memory utilization information.
total-mem
Reports the complete memory utilization information.
threshold
Configures the high and low threshold values for the critical resources.
high
(Optional) Configures the resource high watermark value.
low
(Optional) Configures the resource low watermark value.
threshold-value
Threshold value, in percentage.
Usage Guidelines
Use the resource command to configure parameters for critical resources such as CPU, memory, DS0, and DSP to report the utilization status to external entities using the gateway resources for call handling. You can use the voice class resource-group command to enter voice-class configuration mode and configure resource groups. Each resource group has a unique number that identifies a group of resources to be monitored.
When you configure the high watermark values for any of the monitoring resources, be sure not to use more resources than available on the gateway. The high and low watermark values for threshold only indicate that the gateway might run out of resources soon. However, the gateway must still be able to trigger threshold-based reporting to the routing/monitoring entity.
When you configure the low watermark value for the threshold, be sure not to underutilize the gateway resources.
Examples
The following example shows how to configure CPU to report the utilization information to the external entities:
Router> enable Router# configure terminal Router(config)# voice class resource-group 1 Router(config-class)# resource cpu 1-min-avg threshold high 10 low 2Related Commands
Command
Description
debug rai
Enables debugging for Resource Allocation Indication (RAI).
periodic-report interval
Configures periodic reporting parameters for gateway resource entities.
rai target
Configures the SIP RAI mechanism.
show voice class resource-group
Displays the resource group configuration information for a specific resource group or all resource groups.
voice class resource-group
Enters voice-class configuration mode and assigns an identification tag number for a resource group.
resource threshold
To configure a gateway to report H.323 resource availability to its gatekeeper, use the resource thresholdcommand in gateway configuration mode. To disable gateway resource-level reporting, use the no form of this command.
Syntax Description
all
(Optional) High- and low-parameter settings are applied to all monitored H.323 resources. This is the default condition.
high percentage -value
(Optional) Resource utilization level that triggers a Resource Availability Indicator (RAI) message that indicates that H.323 resource use is high. Enter a number between 1 and 100 that represents the high-resource utilization percentage. A value of 100 specifies high-resource usage when any H.323 resource is unavailable. Default is 90 percent.
low percentage-value
(Optional) Resource utilization level that triggers an RAI message that indicates H.323 resource usage has dropped below the high-usage level. Enter a number between 1 and 100 that represents the acceptable resource utilization percentage. After the gateway sends a high-utilization message, it waits to send the resource recovery message until the resource use drops below the value defined by the low parameter. Default is 90 percent.
Command Default
Reports low resources when 90 percent of resources are in use and reports resource availability when resource use drops below 90 percent.
Command History
Release
Modification
12.0(5)T
This command was introduced on the Cisco AS5300.
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 is not included in this release.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(11)T
This command was integrated into Cisco IOS Release 12.2(11)T. This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release
Usage Guidelines
This command defines the resource load levels that trigger RAI messages. To view the monitored resources, enter the show gateway command.
The monitored H.323 resources include digital signal processor (DSP) channels and DS0s. Use the show call resource voice stats command to see the total amount of resources available for H.323 calls.
NoteThe DS0 resources that are monitored for H.323 calls are limited to the ones that are associated with a voice POTS dial peer.
See the dial-peer configuration commands for details on how to associate a dial peer with a PRI or channel-associated signaling (CAS) group.
When any monitored H.323 resources exceed the threshold level defined by the high parameter, the gateway sends an RAI message to the gatekeeper with the AlmostOutOfResources field flagged. This message reports high resource usage.
When all gateway H.323 resources drop below the level defined by the low parameter, the gateway sends the RAI message to the gatekeeper with the AlmostOutOfResources field cleared.
When a gatekeeper can choose between multiple gateways for call completion, the gatekeeper uses internal priority settings and gateway resource statistics to determine which gateway to use. When all other factors are equal, a gateway that has available resources is chosen over a gateway that has reported limited resources.
resource-pool (mediacard)
To create a Digital Signal Processor (DSP) resource pool on ad-hoc conferencing and transcoding port adapters, use the resource-poolcommand in mediacard configuration mode. To remove the DSP resource pool and release the associated DSP resources, use the no form of this command.
Usage Guidelines
The DSP resource pool identifier should be unique across the same Communication Media Module (CMM). Removing a resource pool may cause the profile using that resource pool to be disabled if it is the last resource pool in the profile.
response (voice)
To use SIP profiles to add, copy, modify, or remove Session Initiation Protocol (SIP) or Session Description Protocol (SDP) header value in a SIP response message, use the responsecommand in voice class configuration mode. To disable the configuration, use the no form of this command.
response option { sdp-header | sip-header } header-name { add | copy | modify | remove } string
no response option { sdp-header | sip-header } header-name { add | copy | modify | remove } string
Syntax Description
option
Response code to be added, copied, modified, or removed.
You can specify one of the following values:
sdp-header
Specifies SDP header.
sip-header
Specifies SIP header.
header-name
SDP or SIP header name.
add
Adds a header.
copy
Copies a header.
modify
Modifies a header.
remove
Removes a header.
string
String to be added as a header.
Usage Guidelines
If there are interoperability issues with Cisco UBE, the Cisco UBE will not be able to work with the default SIP signaling. Hence, you must modify the SIP profiles to add, copy, modify, or remove SIP header values, to enable Cisco UBE to work with SIP signaling.
Use the response command to modify SIP profiles for a response message. You can add, copy, modify, or remove SIP or SDP header values in an outgoing SIP response message.
response (XML application)
To set XML application response parameters, use the response command in XML application configuration mode. To disable response parameter settings, use the no form of this command.
response { formatted | timeout { -1 | seconds } }
no response { formatted | timeout { -1 | seconds } }
Syntax Description
formatted
Response parameters in formatted human readable XML.
timeout
Application specified response timeout.
-1
Enter -1 to indicate no application specified timeout. This is the default timeout setting.
seconds
Number of seconds a response is active before it times out. Valid range includes 0 to 60 seconds.
Command Default
The default for the timeout keyword is -1 indicating not application specified timeout.
Usage Guidelines
The response timeout specified in this command, if other than -1 which is the default, overwrites the timeout value specified in the request (XML transport) command that sets the timeout at the transport level.
The same http transport layer could have multiple applications active at the same time. You can set the timeout for each application individually or have all of the applications to use the same timeout value set at transport layer using the request (XML transport) command in XML transport configuration mode.
response peer-header
To use SIP profiles to copy a peer header value in a SIP response message, use the response peer-header command in voice class configuration mode. To disable the configuration, use the no form of this command.
response { code | any } peer-header sip { sip-req-uri | header-name } copy pattern variable
no response option peer-header sip { sip-req-uri | header-name } copy pattern variable
Syntax Description
code
Response code to be copied. You can specify one of the following values:
- any --Adds, copies, modifies, or removes any response message.
any
Adds, copies, modifies, or removes any response message.
sip
Specifies that the SIP header must be copied from the peer call leg.
sip-req-uri
Specifies the SIP request Uniform Resource Identifier (URI) to be copied from the peer call leg.
header-name
Header name from which the peer header values must be copied.
copy
Copies a header.
pattern
Match pattern.
variable
The destination variable name. The range is from u01 to u99.
Usage Guidelines
If there are interoperability issues with Cisco UBE, the Cisco UBE will not be able to work with the default SIP signaling. Hence, you must modify the SIP profiles to add, copy, modify, or remove SIP or SDP header values, to enable Cisco UBE to work with SIP signaling.
Use the response peer-header command to copy a peer header value in a SIP response message.
response size (XML transport)
To set the response transport fragment size, use the response size command in XML transport configuration mode. To disable the response transport fragment size setting, use the no form of this command.
Usage Guidelines
The fragment size is constrained by the transport type. The CLI help provides input guidelines.
response-timeout
To configure the maximum time to wait for a response from a server, use the response-timeoutcommand in settlement configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
If no response is received within the response-timeout time limit, the current connection ends, and the router attempts to contact the next service point.
Related Commands
Command
Description
connection -timeout
Configures the time for which a connection is maintained after completion of a communication exchange.
customer -id
Identifies a carrier or ISP with a settlement provider.
device -id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max -connection
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
retry -delay
Sets the time between attempts to connect with the settlement provider.
retry -limit
Sets the maximum number of attempts to connect to the provider.
session -timeout
Sets the interval for closing the connection when there is no input or output traffic.
settlement
Enters settlement mode and specifies the attributes specific to a settlement provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown/no shutdown
Deactivates the settlement provider/activates the settlement provider.
type
Configures an SAA-RTR operation type.
url
Specifies the Internet service provider address.
retries (auto-config application)
To set the number of download retry attempts for an auto-configuration application, use the retries command in auto-config application configuration mode. To reset to the default, use the no form of this command.
retry bye
To configure the number of times that a BYE request is retransmitted to the other user agent, use the retry byecommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
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 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400 and Cisco AS5850 in this release.
Related Commands
Command
Description
default
Resets the value of a command to its default.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
sip-ua
Enables the SIP user-agent configuration commands, with which you configure the user agent.
retry cancel
To configure the number of times that a CANCEL request is retransmitted to the other user agent, use the retry cancelcommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
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 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400 and Cisco AS5850 in this release.
Related Commands
Command
Description
default
Resets the value of a command to its default.
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
sip-ua
Enables the sip ua configuration commands, with which you configure the user agent.
retry comet
To configure the number of times that a COMET request is retransmitted to the other user agent, use the retry cometcommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
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, and Cisco AS5400 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Usage Guidelines
COMET, or conditions met, indicates if preconditions for a given call or session have been met. This command is applicable only with calls (other than best-effort) that involve quality of service (QoS).
Use the default number of 10 retries, when possible. Lower values, such as 1, can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example configures aCOMET request to be retransmitted 8 times:
Router(config)# sip-ua Router(config-sip-ua)# retry comet 8Related Commands
Command
Description
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
show sip -ua retry
Displays the SIP retry attempts.
show sip -ua statistics
Displays response, traffic, timer, and retry statistics.
retry interval
To define the time between border element attempts delivery of unacknowledged call-detail-record (CDR) information, use the retry intervalcommand in Annex G neighbor usage configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Use this command to set the interval during which the border element attempts delivery of unacknowledged call-detail-record (CDR) information.
Examples
The following example sets the retry interval to 2700 seconds (45 minutes):
Router(config-nxg-neigh-usg)# retry interval 2700Related Commands
Command
Description
access-policy
Requires that a neighbor be explicitly configured.
inbound ttl
Sets the inbound time-to-live value.
outbound retry-interval
Defines the retry period for attempting to establish the outbound relationship between border elements.
retry window
Defines the total time for which a border element attempts delivery.
service-relationship
Establishes a service relationship between two border elements.
shutdown
Enables or disables the border element.
usage-indication
Enters the mode used to configure optional usage indicators.
retry invite
To configure the number of times that a Session Initiation Protocol (SIP) INVITE request is retransmitted to the other user agent, use the retry invitecommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
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 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Usage Guidelines
To reset this command to the default value, you can also use the default command.
When configuring SIP using SIP user-agent configuration commands such as the retry invite command, the use of the default values for the commands causes the rotary function to not take effect. The rotary function is when you set up more than one VoIP dial peer for the same destination pattern, and the dial peers are assigned to different targets. Assign different targets so that if the call cannot be set up with the first dial peer (preference one), the next dial peer can be tried.
To use the rotary function within SIP, set the retry value for the SIP retry invitecommand to 4 or less.
Related Commands
Command
Description
default
Resets the value of a command to its default.
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
sip-ua
Enables the UA configuration commands, with which you configure the user agent.
retry keepalive (SIP)
To set the retry count for keepalive retransmission, use the retry keepalive command in SIP UA configuration mode. To restore the retry count to the default value for keepalive retransmission, use the no form of this command.
Examples
The following example sets the retry for the keepalive retransmissions to 8:
sip-ua retry keepalive 8Related Commands
Command
Description
busyout monitor keepalive
Selects a voice port or ports to be busied out in cases of a keepalive failure.
keepalive target
Identifies a SIP server that will receive keepalive packets from the SIP gateway.
keepalive trigger
Sets the trigger to the number of Options message requests that must consecutively receive responses from the SIP servers in order to unbusy the voice ports when in the down state.
timers keepalive
Sets the time interval between sending Options message requests when the SIP server is active or down.
retry notify
To configure the number of times that the notify message is retransmitted to the user agent that initiated the transfer or Refer request, use the retry notifycommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB2
This command was implemented on the Cisco AS5850.
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 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Cisco IOS XE Release 2.5
This command was integrated into Cisco IOS XE Release 2.5.
Usage Guidelines
A notify message informs the user agent that initiated the transfer or refer request of the outcome of the Session Initiation Protocol (SIP) transaction.
Use the default number of 10 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example configures anotify message to be retransmitted 10 times:
Router(config)# sip-ua Router(config-sip-ua)# retry notify 10Related Commands
Command
Description
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a Session Initiation Protocol (SIP) INVITE request is retransmitted to the other user agent.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
show sip-ua retry
Displays the SIP retry attempts.
show sip-ua statistics
Displays response, traffic, timer, and retry statistics.
timers notify
Sets the amount of time that the user agent should wait before retransmitting the Notify message.
retry prack
To configure the number of times that the PRACK request is retransmitted to the other user agent, use the retry prackcommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 platforms is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Usage Guidelines
PRACK allows reliable exchanges of Session Initiation Protocol (SIP) provisional responses between SIP endpoints. Use the default number of 10 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example configures aPRACK request to be retransmitted 9 times:
Router(config)# sip-ua Router(config-sip-ua)# retry prack 9Related Commands
Command
Description
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
show sip-ua retry
Displays the SIP retry attempts.
show sip-ua statistics
Displays response, traffic, timer, and retry statistics.
retry refer
To configure the number of times that the Refer request is retransmitted, use the retry refercommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
A Session Initiation Protocol (SIP) Refer request is sent by the originating gateway to the receiving gateway and initiates call forward and call transfer capabilities.
When configuring the retry refer command, use the default number of 10 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the receiving gateway.
retry register
To set the total number of Session Initiation Protocol (SIP) register messages that the gateway should send, use the retry register command in SIP user-agent configuration mode. To reset this number to the default, use the no form of this command.
retry register retries [ exhausted-random-interval minimum minutes maximum minutes ]
no retry register
Syntax Description
retries
Total number of register messages that the gateway should send. The range is from 1 to 10. The default is 6 retries.
exhausted-random-interval
Specifies the register request to be generated within the defined range of time intervals.
minimum minutes
Specifies the minimum time interval range, in minutes, that will be used as the interval before the next registration is sent.
maximum minutes
Specifies the maximum time interval range, in minutes, that will be used as the interval before the next registration is sent.
Command History
Release
Modification
12.2(15)ZJ
This command was introduced.
12.3(4)T
This command was integrated into Cisco IOS Release 12.3(4)T.
12.4(22)T
This command was modified. Support for IPv6 was added.
12.4(22)YB
This command was modified. The exhausted-random-interval keyword was added.
15.0(1)M
This command was integrated into Cisco IOS Release 15.0(1)M.
Usage Guidelines
Use the default number when possible. Lower values such as 1 may lead to the message not being received by the other user agent.
Examples
The following example shows how to configure the gateway to send 9 register messages:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# retry register 9The following example shows how to configure the gateway to send 6 register messages and choose a random number between 2 and 5 as the interval before sending the next registration message:
Router> enable Router# configure terminal Router(config)# sip-ua Router(config-sip-ua)# retry register 6 exhausted-random-interval minimum 2 maximum 5Related Commands
Command
Description
registrar
Enables SIP gateways to register E.164 numbers on behalf of analog telephone voice ports (FXS), IP phone virtual voice ports (EFXS), and SCCP phones with an external SIP proxy or SIP registrar.
timers register
Sets how long the SIP user agent waits before sending register requests.
retry rel1xx
To configure the number of times that the reliable 1xx response is retransmitted to the other user agent, use the retry rel1xxcommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.2(2)XB
This command was introduced.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.
12.2(11)T
This command is supported on the Cisco AS5300, Cisco AS5350, and Cisco AS5400 in this release.
Usage Guidelines
Use the default number of 6 when possible. Lower values such as 1 can lead to an increased chance of the message not being received by the other user agent.
Examples
The following example configures the reliable 1xx response to be retransmitted 7 times:
Router(config)# sip-ua Router(config-sip-ua)# retry rel1xx 7Related Commands
Command
Description
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times the PRACK request is retransmitted.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
show sip-ua retry
Displays the SIP retry attempts.
show sip-ua statistics
Displays response, traffic, timer, and retry statistics.
retry response
To configure the number of times that the response message is retransmitted to the other user agent, use the retry responsecommand in SIP UA configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.1(1)T
This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.2(2)XA
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
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 is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Related Commands
Command
Description
default
Resets the value of a command to its default.
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times the PRACK request is retransmitted.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
sip-ua
Enables the sip-ua configuration commands, with which you configure the user agent.
retry subscribe
To configure the number of times that a SIP SUBSCRIBE message is retransmitted to the other user agent, use the retry subscribe command in SIP UA configuration mode. To reset to the default, use the no form of this command.
retry window
To define the total time for which a border element attempts delivery, use the retry windowcommand in Annex G neighbor usage configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Use this command to set the total time during which a border element attempts delivery of unacknowledged call-detail-record (CDR) information.
Examples
The following example sets the retry window to 15 minutes:
Router(config-nxg-neigh-usg)# retry window 15Related Commands
Command
Description
access-policy
Requires that a neighbor be explicitly configured.
inbound ttl
Sets the inbound time-to-live value.
outbound retry-interval
Defines the retry period for attempting to establish the outbound relationship between border elements.
retry bye
Configures the number of times that a BYE request is retransmitted to the other user agent.
retry cancel
Configures the number of times that a CANCEL request is retransmitted to the other user agent.
retry comet
Configures the number of times that a COMET request is retransmitted to the other user agent.
retry invite
Configures the number of times that a SIP INVITE request is retransmitted to the other user agent.
retry notify
Configures the number of times that the Notify message is retransmitted to the user agent that initiated the transfer or Refer request.
retry prack
Configures the number of times that the PRACK request is retransmitted to the other user agent.
retry rel1xx
Configures the number of times that the reliable 1xx response is retransmitted to the other user agent.
retry response
Configures the number of times that the RESPONSE message is retransmitted to the other user agent.
service-relationship
Establishes a service relationship between two border elements.
shutdown
Enables or disables the border element.
usage-indication
Enters the submode used to configure optional usage indicators.
retry-delay
To set the time between attempts to connect with the settlement provider, use the retry-delay command in settlement configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
After exhausting all service points for the provider, the router is delayed for the specified length of time before resuming connection attempts.
Related Commands
Command
Description
connection -timeout
Configures the time for which a connection is maintained after completion of a communication exchange.
customer -id
Identifies a carrier or ISP with a settlement provider.
device -id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max -connection
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
response -timeout
Configures the maximum time to wait for a response from a server.
retry -limit
Sets the maximum number of attempts to connect to the provider.
session -timeout
Sets the interval for closing the connection when there is no input or output traffic.
settlement
Enters settlement configuration mode and specifies the attributes specific to a settlement provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown/no shutdown
Deactivates the settlement provider/activates the settlement provider.
type
Configures an SAA-RTR operation type.
retry-limit
To set the maximum number of attempts to connect to the provider, use the retry-limit command in settlement configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
If no connection is established after the configured number of retries has been attempted, the router ceases connection attempts. The retry limit number does not count the initial connection attempt. A retry limit of one (default) results in a total of two connection attempts to every service point.
Related Commands
Command
Description
connection -timeout
Configures the time for which a connection is maintained after a communication exchange is complete.
customer -id
Identifies a carrier or ISP with a settlement provider.
device -id
Specifies a gateway associated with a settlement provider.
encryption
Sets the encryption method to be negotiated with the provider.
max -connection
Sets the maximum number of simultaneous connections to be used for communication with a settlement provider.
response -timeout
Configures the maximum time to wait for a response from a server.
retry -delay
Sets the time between attempts to connect with the settlement provider.
session -timeout
Sets the interval for closing the connection when there is no input or output traffic.
settlement
Enters settlement mode and specifies the attributes specific to a settlement provider.
show settlement
Displays the configuration for all settlement server transactions.
shutdown
Brings up the settlement provider.
type
Configures an SAA-RTR operation type.
ring
To set up a distinctive ring for your connected telephones, fax machines, or modems, use the ringcommand in interface configuration mode. To disable the ring, use the no form of this command.
Syntax Description
cadence -number
Number that determines the ringing cadence. Range is from 0 to 2:
- Type 0 is a primary ringing cadence--default ringing cadence for the country your router is in.
- Type 1 is a distinctive ring--0.8 seconds on, 0.4 seconds off, 0.8 seconds on, 0.4 seconds off.
- Type 2 is a distinctive ring--0.4 seconds on, 0.2 seconds off, 0.4 seconds on, 0.2 seconds off, 0.8 seconds on, 4 seconds off.
Usage Guidelines
This command applies to Cisco 800 series routers.
You can specify this command when creating a dial peer. This command does not work if it is not specified within the context of a dial peer. For information on creating a dial peer, see to the Cisco 800 Series Routers Software Configuration Guide.
Related Commands
Command
Description
destination -pattern
Specifies the prefix, the full E.164 telephone number, or an ISDN directory number to be used for a dial peer.
dial -peer voice
Enters dial-peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.
no call -waiting
Disables call waiting.
port (dial -peer)
Enables an interface on a PA-4R-DTR port adapter to operate as a concentrator port.
pots distinctive -ring-guard-time
Specifies a delay during which a telephone port can be rung after a previous call is disconnected (for Cisco 800 series routers).
show dial -peer voice
Displays configuration information and call statistics for dial peers.
ring cadence
To specify the ring cadence for a Foreign Exchange Station (FXS) voice port, use the ring cadence command in voice-port configuration mode. To reset to the default, use the no form of this command.
ring cadence { pattern-number | define pulse interval }
no ring cadence
{ ring cadence external patternXX | define }
{ ring cadence patternXX | define }
Syntax Description
pattern -number
Predefined ring cadence patterns. Each pattern specifies a ring-pulse time and a ring-interval time.
- pattern01 -- 2 seconds on, 4 seconds off
- pattern02 -- 1 second on, 4 seconds off
- pattern03 -- 1.5 seconds on, 3.5 seconds off
- pattern04 -- 1 second on, 2 seconds off
- pattern05 -- 1 second on, 5 seconds off
- pattern06 -- 1 second on, 3 seconds off
- pattern07 -- 0.8 second on, 3.2 seconds off
- pattern08 -- 1.5 seconds on, 3 seconds off
- pattern09 -- 1.2 seconds on, 3.7 seconds off
- pattern09 -- 1.2 seconds on, 4.7 seconds off
- pattern11 -- 0.4 second on, 0.2 second off, 0.4 second on, 2 seconds off
- pattern12 -- 0.4 second on, 0.2 second off, 0.4 second on, 2.6 seconds off
define
User-definable ring cadence pattern. Each number pair specifies one ring-pulse time and one ring-interval time. You must enter numbers in pairs, and you can enter from 1 to 6 pairs. The second number in the last pair that you enter specifies the interval between rings.
pulse
Number (1 or 2 digits) specifying ring-pulse (on) time in hundreds of milliseconds.
Range is from 1 to 50, for pulses of 100 to 5000 ms. For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
interval
Number (1 or 2 digits) specifying ring-interval (off) time in hundreds of milliseconds.
Range is from 1 to 50, for pulses of 100 to 5000 ms. For example: 1 = 100 ms; 10 = 1 s, 40 = 4 s.
Command History
Release
Modification
11.3(1)MA
This command was introduced on the Cisco MC3810.
12.0(7)XK
This command was implemented on the Cisco 2600 series and Cisco 3600 series. The patternXX keyword was added.
12.1(2)T
This command was integrated into Cisco IOS Release 12.1(2)T.
15.0(1)M
This command was modified. The external keyword was added to specify the ring pattern of external calls.
Usage Guidelines
To specify the ring pattern for external calls, use the ring cadence external command. It is supported only in STCAPP. To specify the ring cadence for internal calls, use the existing ring cadence command. The syntax for the ring cadence external command is the same as for the ring cadence command.
The patternXX keyword provides preset ring cadence patterns for use on any platform. The define keyword allows you to create a custom ring cadence. On the Cisco 2600 and Cisco 3600 series routers, only one or two pairs of digits can be entered under the define keyword.
ring frequency
To specify the ring frequency for a specified Foreign Exchange Station (FXS) voice port, use the ring frequencycommand in voice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Use this command to select a specific ring frequency for an FXS voice port. Use the no form of this command to reset the default value. The ring frequency you select must match the connected equipment. If set incorrectly, the attached phone might not ring or might buzz. In addition, the ring frequency is usually country-dependent. You should take into account the appropriate ring frequency for your area before configuring this command.
This command does not affect ringback, which is the ringing a user hears when placing a remote call.
ring number
To specify the number of rings for a specified Foreign Exchange Office (FXO) voice port, use the ring numbercommand in voice port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Use this command to set the maximum number of rings to be detected before answering a call over an FXO voice port. Use the no form of this command to reset the default value, which is one ring.
Normally, this command should be set to the default so that incoming calls are answered quickly. If you have other equipment available on the line to answer incoming calls, you might want to set the value higher to give the equipment sufficient time to respond. In that case, the FXO interface would answer if the equipment online did not answer the incoming call in the configured number of rings.
This command is not applicable to Foreign Exchange Station (FXS) or E&M interfaces because they do not receive ringing on incoming calls.
ringing-timeout
To define the timeout period for the SCCP telephony control (STC) application feature call back, use the ringing-timeoutcommand in STC application feature callback configuration mode. To return to the default timeout period, use the no form of this command.
Usage Guidelines
This command changes the timeout period of the ringing timer from the default of 30 seconds to the specified value.
The ringing timer specifies the number of seconds during which the calling device that is in a Callback on Busy condition can receive a Callback Ringing and after which, if the calling device does not answer, the CallBack on Busy condition is cancelled.
roaming (dial peer)
To enable roaming capability for a dial peer, use the roaming command in dial-peer configuration mode. To disable roaming capability, use the no form of this command.
Usage Guidelines
Use this command to enable roaming capability of a dial peer if that dial peer can terminate roaming calls. If a dial peer is dedicated to local calls only, disable roaming capability.
The roaming dial peer must work with a roaming service provider. If the dial peer allows a roaming user to go through and the service provider is not roaming-enabled, the call fails.
Examples
The following example enables roaming capability for a dial peer:
dial-peer voice 10 voip roamingRelated Commands
Command
Description
roaming (settlement)
Enables the roaming capability for a settlement provider.
settle-call
Limits the dial peer to using only the specific clearinghouse identified by the specified >provider ->number .
settlement roam-pattern
Configures a pattern to match against when determining roaming.
roaming (settlement)
To enable roaming capability for a settlement provider, use the roaming command in settlement configuration mode. To disable roaming capability, use the no form of this command.
Usage Guidelines
Enable roaming capability of a settlement provider if that provider can authenticate a roaming user and route roaming calls.
A roaming call is successful only if both the settlement provider and the outbound dial peer for that call are roaming-enabled.
Examples
The following example enables roaming capability for a settlement provider:
settlement 0 roamingRelated Commands
Command
Description
roaming (dial-peer mode)
Enables the roaming capability for the dial peer.
settle-call
Limits the dial peer to using only the specific clearinghouse identified by the specified >provider ->number .
settlement roam-pattern
Configures a pattern to match against when determining roaming.
rrq dynamic-prefixes-accept
To enable processing of additive registration request (RRQ) RAS messages and dynamic prefixes on the gatekeeper, use the rrq dynamic-prefixes-accept command in gatekeeper configuration mode. To disable processing of additive RRQ messages and dynamic prefixes, use the no form of this command.
Command Default
In Cisco IOS Release 12.2(15)T, the default was set to enabled. In Cisco IOS Release 12.3(3), the default is set to disabled.
Usage Guidelines
In Cisco IOS Release 12.2(15)T, the default for the rrq dynamic-prefixes-accept command was set to enabled so that the gatekeeper automatically received dynamic prefixes in additive RRQ messages from the gateway. Beginning in Cisco IOS Release 12.3(3), the default is set to disabled, and you must specify the command to enable the functionality.
rsvp
To enable RSVP support on a transcoding or MTP device, use the rsvp command in DSP farm profile configuration mode. To disable RSVP support, use the no form of this command.
Usage Guidelines
This command enables a transcoder or MTP device to register as RSVP-capable with Cisco Unified CallManager. The SCCP device acts as an RSVP agent under the control of Cisco Unified CallManager. To support RSVP, you must also enable the codec pass-through command.
NoteThis command is not supported in conferencing profiles.
NoteExamples
The following example enables RSVP support on the transcoding device defined by profile 200:
Router(config)# dspfarm profile 200 transcode Router(config-dspfarm-profile)# rsvp Router(config-dspfarm-profile)# codec pass-throughRelated Commands
Command
Description
codec (DSP Farm profile)
Specifies the codecs supported by a DSP farm profile.
debug call rsvp-sync events
Displays events that occur during RSVP setup.
dspfarm profile
Enters DSP farm profile configuration mode and defines a profile for DSP farm services.
show sccp connections rsvp
Displays information about active SCCP connections that use RSVP.
rtcp keepalive
To configure RTP Control Protocol (RTCP) keepalive report generation and generate RTCP keepalive packets, use the rtcp keepalivecommand in voice service configuration mode. To disable the configuration, use the no form of this command.
Usage Guidelines
Use this command to configure RTCP keepalive report generation and generate RTCP keepalive packets. The no form of the command restores the default behavior.
Examples
The following example shows how to configure RTCP keepalive report generation and generate RTCP keepalive packets:
Router> enable Router# configure terminal Router(config) voice service voip Router(conf-voi-serv)# rtcp keepalivertp payload-type
To identify the payload type of a Real-Time Transport Protocol (RTP) packet, use the rtp payload-type command in dial peer voice configuration mode. To remove the RTP payload type, use the no form of this command.
rtp payload-type { cisco-cas-payload number | cisco-clear-channel number | cisco-codec-aacld number | cisco-codec-fax-ack number | cisco-codec-fax-ind number | cisco-codec-gsmamrnb number | cisco-codec-ilbc number | cisco-codec-isac number | cisco-codec-video-h263+ number | cisco-codec-video-h264 number | cisco-fax-relay number | cisco-pcm-switch-over-alaw number | cisco-pcm-switch-over-ulaw number | cisco-rtp-dtmf-relay number | lmr-tone number | nse number | nte number | nte-tone number } [ comfort-noise { 13 | 19 } ]
no rtp payload-type { cisco-cas-payload number | cisco-clear-channel number | cisco-codec-fax-ack number | cisco-codec-fax-ind number | cisco-codec-gsmamrnb number | cisco-codec-ilbc number | cisco-codec-video-h263+ number | cisco-codec-video-h264 number | cisco-fax-relay number | cisco-pcm-switch-over-alaw number | cisco-pcm-switch-over-ulaw number | cisco-rtp-dtmf-relay number | lmr-tone number | nse number | nte number | nte-tone number } [ comfort-noise { 13 | 19 } ]
Syntax Description
cisco-cas-payload number
Cisco channel-associated signaling (CAS) RTP payload. Range: 96 to 127. Default: 123.
cisco-clear-channel number
Cisco clear-channel RTP payload. Range: 96 to 127. Default: 125.
cisco-codec-aacld number
Cisco MPEG-4 Advanced Audio Codec - Low Delay (AAC_LD) codec. Range: 96 to 127. Default: 114.
cisco-codec-fax-ack number
Cisco codec fax acknowledge. Range: 96 to 127. Default: 97.
cisco-codec-fax-ind number
Cisco codec fax indication. Range: 96 to 127. Default: 96.
cisco-codec-gsmamrnb number
Cisco Global System for Mobile Adaptive Multi-Rate Narrow Band (GSMAMR-NB) codec. Range: 96 to 127. Default: 117.
cisco-codec-ilbc number
Cisco internet Low Bitrate Codec (iLBC) codec. Range: 96 to 127. Default: 116.
cisco-codec-isac number
Cisco internet Speech Audio Codec (iSAC) codec. Range: 96 to 127. Default: 124.
cisco -codec-video-h263+ number
RTP video codec H.263+ payload type. Range: 96 to 127. Default: 118.
cisco -codec-video-h264 number
RTP video codec H.264 payload type. Range: 96 to 127. Default: 119.
cisco-fax-relay number
Cisco fax relay. Range: 96 to 127. Default: 122.
cisco-pcm-switch-over-alaw number
Cisco RTP pulse code modulation (PCM) codec switch over indication (a-law). Default: 8.
cisco-pcm-switch-over-ulaw number
Cisco RTP PCM codec switch over indication (mu-law). Default: 0.
cisco-rtp-dtmf-relay number
Cisco RTP dual-tone multifrequency (DTMF) relay. Range: 96 to 127. Default: 121.
lmr-tone number
LMR payload type. Range: 96 to 127. Default: 0. The default value is set by the no rtp payload-type lmr-tone command.
nse number
A named signaling event (NSE). Range: 96 to 117. Default: 100.
nte number
A named telephone event (NTE). Range: 96 to 127. Default: 101.
nte-tone number
RFC-2833 tone payload type. Range 96 to 127. Default: 101.
comfort-noise 13 19
(Optional) RTP payload type of comfort noise. The July 2001 draft entitled RTP Payload for Comfort Noise , from the Internet Engineering Task Force (IETF) Audio/Video Transport (AVT) working group, designates 13 as the payload type for comfort noise. If you are connecting to a gateway that complies with the RTP Payload for Comfort Noise draft, use 13. Use 19 only if you are connecting to older Cisco gateways that use DSPware before version 3.4.32.
Note This command option is not available on the Cisco AS5400 running NextPort digital signal processors (DSPs). This command option is available on the Cisco AS5400 only if the platform has a high-density packet voice/fax feature card (AS5X-FC) with one or more AS5X-PVDM2-64 DSP modules installed. This support was added in Cisco IOS Release 12.4(4)XC, and integrated into Release 12.4(9)T, and later 12.4T releases.
Command History
Release
Modification
12.2(2)T
This command was introduced.
12.2(2)XB
This command was modified. The nte and comfort - noise keywords were added.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(8)T
This command was integrated into Cisco IOS Release 12.2(8)T.
12.4(4)XC
This command was modified. The cisco-codec-gsmamrnb keyword was added.
12.4(9)T
This command was integrated into Cisco IOS Release 12.4(9)T.
12.4(11)T
This command was modified. The cisco-codec-ilbc, cisco-codec-video-h263+, and cisco-codec-video-h264 keywords were added.
12.4(15)XY
This command was modified. The lmr-tone and nte-tone keywords were added.
12.4(20)T
This command was integrated into Cisco IOS Release 12.4(20)T.
IOS Release XE 2.5
This command was integrated into Cisco IOS XE Release 2.5.
15.1(1)T
This command was modified. The cisco-codec-isac keyword was added.
Usage Guidelines
Use this command to identify the payload type of an RTP. Use this command after the dtmf-relaycommand is used to choose the NTE method of DTMF relay for a Session Initiation Protocol (SIP) call.
Configured payload types of NSE and NTE exclude certain values that have been previously hard-coded with Cisco-proprietary meanings. Do not use the following numbers, which have preassigned values: 96, 97, 100, 117, 121 to 123, and 125 to 127.
Use of these values results in an error message when the command is entered. You must first reassign the value in use to a different unassigned number, for example:
rtp payload-type cisco-codec-ilbc 100 ERROR: value 100 in use! rtp payload-type nse 105 rtp payload-type cisco-codec-ilbc 100Examples
The following example shows how to identify the RTP payload type as GSMAMR-NB115:
Router(config-dial-peer)# rtp payload-type cisco-codec-gsmamrnb 115The following example shows how to identify the RTP payload type as NTE 99:
Router(config-dial-peer)# rtp payload-type nte 99The following example shows how to identify the RTP payload type for the iLBC as 100:
Router(config-dial-peer)# rtp payload-type cisco-codec-ilbc 100rtp send-recv
To configure a Cisco IOS Session Initiation Protocol (SIP) gateway to establish a bidirectional voice path as soon as it receives a SIP 183 PROGRESS message with Session Description Protocol (SDP), use the rtp send-recv command in voice service SIP configuration mode. To configure the gateway to establish a backward-only media cut-through voice path upon receipt of a 183 PROGRESS message with SDP that persists until the call progresses to the connect state, use the no form of this command.
Command Default
A bidirectional voice path is established upon receipt of a 183 PROGRESS message with SDP.
Usage Guidelines
The default behavior on a Cisco IOS SIP gateway is to establish a bidirectional voice path from the moment it receives a SIP 183 PROGRESS message with SDP. However, this can result in clipping on some voice platforms if both parties send audio at the same time, such as during a call setup process when interactive voice response (IVR) and a caller both speak simultaneously. To establish the voice path in the backward direction only until the call is connected, use the no rtp send-recv command in voice service SIP configuration mode.
A backward-only voice path operates only during the connection attempt--once a call is connected, the voice path automatically converts to bidirectional sending and receiving of Real-Time Transport Protocol (RTP) packets and RTP control packets (RTCPs). However, if the no rtp send-recvcommand is configured on a SIP gateway, no inband or RFC 2833-based dual tone multifrequency (DTMF) digits can be sent in the forward direction until after the call is connected and the bidirectional voice path is established.
rtp-ssrc multiplex
To multiplex Real-Time Transport Control Protocol (RTCP) packets with RTP packets and to send multiple synchronization source in RTP headers (SSRCs) in a RTP session, use the rtp-ssrc multiplexcommand in voice service or dial peer voice configuration mode. To disable the configuration, use the no form of this command.
Syntax Available Under Voice Service Configuration Mode
rtp-ssrc multiplex
no rtp-ssrc multiplex
Syntax Available Under Dial Peer Voice Configuration Mode
rtp-ssrc multiplex [system]
no rtp-ssrc multiplex [system]
Command Default
Under voice service configuration mode, the rtp-ssrc multiplex command is not enabled and hence there is no interoperation with Cisco TelePresence System (CTS).
At the dial-peer level, the rtp-ssrc multiplex command uses the global configuration level settings.
Command Modes
Voice service configuration (conf-voi-serv)
Dial peer voice configuration (config-dial-peer)rtsp client session history duration
To specify how long to keep Real Time Streaming Protocol (RTSP) client history records in memory, use the rtsp client session history duration command in global configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1750 and Cisco 1751. This release does not support any other Cisco platforms.
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 is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Examples
The following example sets the duration for the RTSP session history to 500 minutes:
rtsp client session history duration 500Related Commands
Command
Description
call application voice load
Allows reload of an application that was loaded via the MGCP scripting package.
rtsp client session history records
Specifies the number of RTSP client session history records kept during the session.
show call application voice
Displays all TCL or MGCP scripts that are loaded.
show rtsp client session
Displays cumulative information about the RTSP session records.
rtsp client rtpsetup enable
To configure a router to send the IP address in a Real Time Streaming Protocol (RTSP) setup message, use the rtsp client rtpsetup enable command in global configuration mode. To disable the configuration, use the no form of this command.
rtsp client session history records
To configure the number of records to keep in the Real Time Streaming Protocol (RTSP) client session history, use the rtsp client session history records command in global configuration mode. To reset to the default, use the no form of this command.
Command History
Release
Modification
12.1(3)T
This command was introduced on the Cisco AS5300.
12.1(5)T
This command was implemented on the Cisco AS5800.
12.1(5)XM2
This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB1
This command was implemented on the Cisco AS5850.
12.2(4)XM
This command was implemented on the Cisco 1750 and Cisco 1751. This release does not support any other Cisco platforms.
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 is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
Examples
The following example specifies that a total of 500 records are to be kept in the RTSP client history:
rtsp client session history records 500Related Commands
Command
Description
call application voice load
Allows reload of an application that was loaded via the MGCP scripting package.
rtsp client session history duration
Specifies the how long the RTSP is kept during the session.
show call application voice
Displays all Tcl or MGCP scripts that are loaded.
rtsp client timeout connect
To set the number of seconds allowed for the router to establish a TCP connection to a Real -Time Streaming Protocol (RTSP) server, use the rtsp client timeout connectcommand in global configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
This command determines when the router abandons its attempt to connect to an RTSP server and declares a timeout error, if a connection cannot be established after the specified number of seconds.
rtsp client timeout message
To set the number of seconds that the router waits for a response from a Real -Time Streaming Protocol (RTSP) server, use the rtsp client timeout messagecommand in global configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
This command sets how long the router waits for the RTSP server to respond to a request before declaring a timeout error.
rule (ENUM configuration)
To define a rule for an ENUM match table, use the rule command in ENUM configuration mode. To delete the rule, use the noform of this command.
rule rule-number preference /match-pattern /replacement-rule /domain-name
rule rule-number preference /match-pattern /replacement-rule /domain-name
Syntax Description
rule -number
Assigns an identification number to the rule. Range is from 1 to 2147483647.
preference
Assigns a preference value to the rule. Range is from 1 to 2147483647. Lower values have higher preference.
/ match -pattern
Stream editor (SED) expression used to match incoming call information. The slash "/" is a delimiter in the pattern.
/ replacement -rule
SED expression used to repla ce match-pattern in the call information. The slash "/" is a delimiter in the pattern.
/ domain -name
Domain name to be used while the query to the DNS server is sent.
Usage Guidelines
The table below shows examples of match patterns, input strings, and result strings for the rule (voice translation-rule) command.
Table 1 Match Patterns, Input Strings and Result Strings Match Pattern
Replacement Pattern
Input String
Result String
Description
/^.*/
//
4085550100
--
Any string to null string.
/^456\(.*\)/
/555\1/
5550100
5550100
Match from the beginning of the input string.
/\(^...\)456\(...\)/
/\1555\2/
408555010
4085550100
Match from the middle of the input string.
/\(.*\)0100/
/\0199/
4085550100
4085550199
Match from the end of the input string.
/^1#\(.*\)/
/\1/
1#2345
2345
Replace match string with null string.
/^408...\(8333\)/
/555\1/
4085550100
5550100
Match multiple patterns.
Rules are entered in any order, but their preference number determines the sequence in which they are used for matching against the input string, which is a called number. A lower preference number is used before a higher preference number.
If a match is found, the input string is modified according to the replacement rule, and the E.164 domain name is attached to the modified number. This longer number is sent to a Domain Name System (DNS) server to determine a destination for the call. The server returns one or more URLs as possible destinations. The originating gateway tries to place the call using each URL in order of preference. If a call cannot be completed using any of the URLs, the call is disconnected.
Examples
The following example defines ENUM rule number 3 with preference 2. The beginning of the call string is checked for digits 9011; when a match is found, 9011 is replaced with 1408 and the call is sent out as an e164.arpa number.
Router(config)# voice enum-match-table number Router(config-enum)# rule 3 2 /^9011\(.*\)//+1408\1/ arparule (voice translation-rule)
To define a translation rule, use the rule command in voice translation-rule configuration mode. To delete the translation rule, use the noform of this command.
Match and Replace Rule
rule precedence /match-pattern/ /replace-pattern/ [ type match-type replace-type [ plan { match-type replace-type } ] ]
no rule precedence
Reject Rule
rule precedence reject /match-pattern/ { type match-type [ plan match-type ] }
no rule precedence
Syntax Description
precedence
Priority of the translation rule. Range is from 1 to 15.
/ match -pattern /
Stream editor (SED) expression used to match incoming call information. The slash ‘/’ is a delimiter in the pattern.
/ replace -pattern /
SED expression used to replace the match pattern in the call information. The slash ‘/’ is a delimiter in the pattern.
type match -type replace-type
(Optional) Number type of the call. Valid values for the match-type argument are as follows:
- abbreviated --Abbreviated representation of the complete number as supported by this network.
- any --Any type of called number.
- international --Number called to reach a subscriber in another country.
- national --Number called to reach a subscriber in the same country, but outside the local network.
- network --Administrative or service number specific to the serving network.
- reserved --Reserved for extension.subscriber--Number called to reach a subscriber in the same local network.
- unknown --Number of a type that is unknown by the network.
Valid values for the replace-type argument are as follows:
type match -type replace-type(continued)
plan match -type replace-type
(Optional) Numbering plan of the call. Valid values for the match-type argument are as follows:
- any --Any type of dialed number.
- data
- ermes
- isdn
- national --Number called to reach a subscriber in the same country, but outside the local network.
- private
- reserved --Reserved for extension.
- telex
- unknown --Number of a type that is unknown by the network.
Valid values for the replace-type argument are as follows:
reject
The match pattern of a translation rule is used for call-reject purposes.
Usage Guidelines
NoteUse this command in conjunction after the voice translation-rule command. An earlier version of this command uses the same name but is used after the translation-rule command and has a slightly different command syntax. In the older version, you cannot use the square brackets when you are entering command syntax. They appear in the syntax only to indicate optional parameters, but are not accepted as delimiters in actual command entries. In the newer version, you can use the square brackets as delimiters. Going forward, we recommend that you use this newer version to define rules for call matching. Eventually, the translation-rulecommand will not be supported.
A translation rule applies to a calling party number (automatic number identification [ANI]) or a called party number (dialed number identification service [DNIS]) for incoming, outgoing, and redirected calls within Cisco H.323 voice-enabled gateways.
Number translation occurs several times during the call routing process. In both the originating and terminating gateways, the incoming call is translated before an inbound dial peer is matched, before an outbound dial peer is matched, and before a call request is set up. Your dial plan should account for these translation steps when translation rules are defined.
The table below shows examples of match patterns, input strings, and result strings for the rule (voice translation-rule) command.
Table 2 Match Patterns, Input Strings and Result Strings Match Pattern
Replacement Pattern
Input String
Result String
Description
/^.*/
//
4085550100
Any string to null string.
//
//
4085550100
4085550100
Match any string but no replacement. Use this to manipulate the call plan or call type.
/\(^...\)456\(...\)/
/\1555\2/
4084560177
4085550177
Match from the middle of the input string.
/\(.*\)0120/
/\10155/
4081110120
4081110155
Match from the end of the input string.
/^1#\(.*\)/
/\1/
1#2345
2345
Replace match string with null string.
/^408...\(8333\)/
/555\1/
4087770100
5550100
Match multiple patterns.
/1234/
/00&00/
5550100
55500010000
Match the substring.
/1234/
/00\000/
5550100
55500010000
Match the substring (same as &).
The software verifies that a replacement pattern is in a valid E.164 format that can include the permitted special characters. If the format is not valid, the expression is treated as an unrecognized command.
The number type and calling plan are optional parameters for matching a call. If either parameter is defined, the call is checked against the match pattern and the selected type or plan value. If the call matches all the conditions, the call is accepted for additional processing, such as number translation.
Several rules may be grouped together into a translation rule, which gives a name to the rule set. A translation rule may contain up to 15 rules. All calls that refer to this translation rule are translated against this set of criteria.
The precedence value of each rule may be used in a different order than that in which they were typed into the set. Each rule’s precedence value specifies the priority order in which the rules are to be used. For example, rule 3 may be entered before rule 1, but the software uses rule 1 before rule 3.
The software supports up to 128 translation rules. A translation profile collects and identifies a set of these translation rules for translating called, calling, and redirected numbers. A translation profile is referenced by trunk groups, source IP groups, voice ports, dial peers, and interfaces for handling call translation.
Examples
The following example applies a translation rule. If a called number starts with 5550105 or 70105, translation rule 21 uses the rule command to forward the number to 14085550105 instead.
Router(config)# voice translation-rule 21 Router(cfg-translation-rule)# rule 1 /^5550105/ /14085550105/ Router(cfg-translation-rule)# rule 2 /^70105/ /14085550105/In the next example, if a called number is either 14085550105 or 014085550105, after the execution of translation rule 345, the forwarding digits are 50105. If the match type is configured and the type is not "unknown," dial-peer matching is required to match the input string numbering type.
Router(config)# voice translation-rule 345 Router(cfg-translation-rule)# rule 1 /^14085550105/ /50105/ plan any national Router(cfg-translation-rule)# rule 2 /^014085550105/ /50105/ plan any national
