Table Of Contents
session
session group
session protocol (dial peer)
session protocol (Voice over Frame Relay)
session protocol aal2
session protocol multicast
session start
session target (MMoIP dial peer)
session target (POTS dial peer)
session target (VoATM dial peer)
session target (VoFR dial peer)
session target (VoIP dial peer)
session transport
session transport (H.323 voice-service)
session transport (SIP)
session-set
set
set http client cache stale
set pstn-cause
set sip-status
settle-call
settlement
settlement roam-pattern
sgcp
sgcp call-agent
sgcp graceful-shutdown
sgcp max-waiting-delay
sgcp modem passthru
sgcp quarantine-buffer disable
sgcp request retries
sgcp request timeout
sgcp restart
sgcp retransmit timer
sgcp timer
sgcp tse payload
session
To associate a transport session with a specified session group, use the session command in backhaul session manager configuration mode. To delete the session, use the no form of this command.
session group group-name remote-ip remote-port local-ip local-port priority
no session group group-name remote-ip remote-port local-ip local-port priority
Syntax Description
group-name
|
Session-group name.
|
remote-ip
|
Remote IP address.
|
remote-port
|
Remote port number. Range is from 1024 to 9999.
|
local-ip
|
Local IP address.
|
local-port
|
Local port number. Range is from 1024 to 9999.
|
priority
|
Priority of the session-group. Range is from 0 to 9999; 0 is the highest priority.
|
Command Default
No default behavior or values
Command Modes
Backhaul session manager configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
12.2(4)T
|
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
|
12.2(2)XB
|
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 implemented on the Cisco IAD2420 series. Support for the Cisco AS5350 and Cisco AS5400 and Cisco AS5850 is not included in this release.
|
12.2(11)T
|
This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
|
Usage Guidelines
It is assumed that the server is located on a remote machine.
Examples
The following example associates a transport session with the session group "group5" and specifies the parameters:
Router(config-bsm)# session group group5 172.13.2.72 5555 172.18.72.198 5555 1
session group
To associate a transport session with a specified session group, use the session group command in backhaul session-manager configuration mode. To delete the session, use the no form of this command.
session group group-name remote-ip remote-port local-ip local-port priority
no session group group-name remote-ip remote-port local-ip local-port priority
Syntax Description
group-name
|
Session-group name.
|
remote-ip
|
Remote IP address.
|
remote-port
|
Remote port number. Range is from 1024 to 9999.
|
local-ip
|
Local IP address.
|
local-port
|
Local port number. Range is from 1024 to 9999.
|
priority
|
Priority of the session group. Range is from 0 to 9999; 0 has the highest priority.
|
Command Default
No default behavior or values.
Command Modes
Backhaul session-manager configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
12.2(2)T
|
This command was implemented on the Cisco 7200 series.
|
12.2(4)T
|
This command was implemented on the following platforms: Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
|
12.2(2)XB1
|
This command was implemented on the Cisco AS5850.
|
12.2(8)T
|
This command was implemented on the Cisco IAD2420 series.
|
Usage Guidelines
The Cisco VSC3000 server is assumed to be located on a remote machine.
Examples
The following example associates a transport session with the session group named "group5" and specifies the keywords described above:
session group group5 172.16.2.72 5555 192.168.72.198 5555 1
session protocol (dial peer)
To specify a session protocol for calls between local and remote routers using the packet network, use the session protocol command in dial peer configuration mode. To reset to the default, use the no form of this command.
session protocol {aal2-trunk | cisco | sipv2 | smtp}
no session protocol
Syntax Description
aal2-trunk
|
Dial peer uses ATM adaptation layer 2 (AAL2) nonswitched trunk session protocol.
|
cisco
|
Dial peer uses the proprietary Cisco VoIP session protocol.
|
sipv2
|
Dial peer uses the Internet Engineering Task Force (IETF) Session Initiation Protocol (SIP). Use this keyword with the SIP option.
|
smtp
|
Dial peer uses Simple Mail Transfer Protocol (SMTP) session protocol.
|
Command Default
No default behaviors or values
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced for VoIP peers on the Cisco 3600 series.
|
12.0(3)XG
|
This command was modified to support VoFR) dial peers.
|
12.0(4)XJ
|
This command was modified for store-and-forward fax on the Cisco AS5300.
|
12.1(1)XA
|
This command was implemented for VoATM dial peers on the Cisco MC3810. The aal2-trunk keyword was added.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T. The sipv2 keyword was added.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.2(2)T
|
This command was implemented on the Cisco 7200 series.
|
12.2(4)T
|
This command was implemented on the Cisco 1750.
|
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 was implemented on the Cisco 7200 series. Supported for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release. The aal2-trunk and smtp keywords are not supported on the Cisco 7200 series in this release.
|
12.2(11)T
|
This command is supported on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
|
Usage Guidelines
The cisco keyword is applicable only to VoIP on the Cisco 1750, Cisco 1751, Cisco 3600 series, and Cisco 7200 series routers.
The aal2-trunk keyword is applicable only to VoATM on the Cisco 7200 series router.
This command applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example shows that AAL2 trunking has been configured as the session protocol:
session protocol aal2-trunk
The following example shows that Cisco session protocol has been configured as the session protocol:
The following example shows that a VoIP dial peer for SIP has been configured as the session protocol for VoIP call signaling:
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial peer configuration mode and specifies the method of voice-related encapsulation.
|
session target (VoIP)
|
Configures a network-specific address for a dial peer.
|
session protocol (Voice over Frame Relay)
To establish a Voice over Frame Relay protocol for calls between the local and remote routers via the packet network, use the session protocol command in dial peer configuration mode. To reset to the default, use the no form of this command.
session protocol {cisco-switched | frf11-trunk}
no session protocol
Syntax Description
cisco-switched
|
Proprietary Cisco VoFR session protocol. (This is the only valid session protocol for the Cisco 7200 series.)
|
frf11-trunk
|
FRF.11 session protocol.
|
Command Default
cisco-switched
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced for VoIP.
|
12.0(3)XG
|
This command was modified to support VoFR on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco MC3810.
|
12.0(4)T
|
The cisco-switched and frf11-trunk keywords were added for VoFR dial peers.
|
Usage Guidelines
For Cisco-to-Cisco dial peer connections, Cisco recommends that you use the default session protocol because of the advantages it offers over a pure FRF.11 implementation. When connecting to FRF.11-compliant equipment from other vendors, use the FRF.11session protocol.
Note When using the FRF.11 session protocol, you must also use the called-number command.
Examples
The following example configures the FRF.11 session protocol for VoFR dial peer 200:
session protocol frf11-trunk
Related Commands
Command
|
Description
|
called-number (dial peer)
|
Enables an incoming VoFR call leg to get bridged to the correct POTS call leg when using a static FRF.11 trunk connection.
|
codec (dial peer)
|
Specifies the voice coder rate of speech for a Voice over Frame Relay dial peer.
|
cptone
|
Specifies a regional analog voice interface-related tone, ring, and cadence setting.
|
destination-pattern
|
Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.
|
dtmf-relay (Voice over Frame Relay)
|
Enables the generation of FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred order of a dial peer within a rotary hunt group.
|
session target
|
Specifies a network-specific address for a specified dial peer or destination gatekeeper.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
session protocol aal2
To enter voice-service-session configuration mode and specify ATM adaptation layer 2 (AAL2) trunking, use the session protocol aal2 command in voice-service configuration mode.
session protocol aal2
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values.
Command Modes
Voice-service configuration
Command History
Release
|
Modification
|
12.1(1)XA
|
This command was introduced on the Cisco MC3810.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.2(2)T
|
This command was implemented on the Cisco 7200 series.
|
Usage Guidelines
This command applies to VoATM on theCisco 7200 series router.
In the voice-service-session configuration mode for AAL2, you can configure only AAL2 features, such as call admission control and subcell multiplexing.
Examples
The following example accesses voice-service-session configuration mode, beginning in global configuration mode:
session protocol multicast
To set the session protocol as multicast, use the session protocol multicast command in dial peer configuration mode. To reset to the default protocol, use the no version of this command.
session protocol multicast
no session protocol multicast
Syntax Description
This command has no arguments or keywords.
Command Default
Default session protocol: Cisco.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
12.1(2)XH
|
This command was introduced for the Cisco Hoot and Holler over IP application on the Cisco 2600 series and Cisco 3600 series.
|
12.1(3)T
|
This command was integrated into Cisco IOS Release 12.1(3)T.
|
12.2(8)T
|
This command was implemented on the Cisco 1750 and Cisco 1751.
|
Usage Guidelines
Use this command for voice conferencing in a hoot and holler networking implementation. This command allows more than two ports to join the same session simultaneously.
Examples
The following example shows the use of the session protocol multicast dial peer configuration command in context with its accompanying commands:
session protocol multicast
session target ipv4:237.111.0.111:22222
Related Commands
Command
|
Description
|
session target ipv4
|
Assigns the session target for voice-multicasting dial peers.
|
session start
To start a new instance (session) of a Tcl IVR 2.0 application, use the session start command in application configuration mode. To stop the session and remove the configuration, use the no form of this command.
session start instance-name application-name
no session start instance-name
Syntax Description
instance-name
|
Alphanumeric label that uniquely identifies this application instance.
|
application-name
|
Name of the Tcl application. This is the name of the application that was assigned with the service command.
|
Command Default
No default behavior or values
Command Modes
Application configuration
Command History
Release
|
Modification
|
12.3(14)T
|
This command was introduced to replace the call application session start (global configuration) command.
|
Usage Guidelines
•This command starts a new session, or instance, of a Tcl IVR 2.0 application. It cannot start a session for a VoiceXML application because Cisco IOS software cannot start a VoiceXML application without an active call leg.
•You can start an application instance only after the Tcl application is loaded onto the gateway with the service command.
•If this command is used, the session restarts if the gateway reboots.
•If the application session stops running, it does not restart unless the gateway reboots. A Tcl script might intentionally stop running by executing a "call close" command for example, or it might fail because of a script error.
•You can start multiple instances of the same application by using different instance names.
Examples
The following example starts a session named my_instance for the application named demo:
application
session start my_instance demo
The following example starts another session for the application named demo:
session start my_instance2 demo
Related Commands
Command
|
Description
|
call application session start (global configuration)
|
Starts a new instance (session) of a Tcl IVR 2.0 application.
|
service
|
Loads a specific, standalone application on a dial peer.
|
show call application services registry
|
Displays a one-line summary of all registered services.
|
show call application sessions
|
Displays summary or detailed information about voice application sessions.
|
session target (MMoIP dial peer)
To designate an e-mail address to receive T.37 store-and-forward fax calls from a Multimedia Mail over IP (MMoIP) dial peer, use the session target command in dial peer configuration mode. To remove the target address, use the no form of this command.
session target mailto:{name | $d$ | $m$ | $e$}[@domain-name]
no session target
Syntax Description
mailto:
|
Matching calls are passed to the network using Simple Mail Transfer Protocol (SMTP) or Extended Simple Mail Transfer Protocol (ESMTP).
|
name
|
String that can be an e-mail address, name, or mailing list alias.
|
$d$
|
Macro that is replaced by the destination pattern of the gateway access number, which is the called number or dialed number identification service (DNIS) number.
|
$m$
|
Macro that is replaced by the redirecting dialed number (RDNIS) if present; otherwise, it is replaced by the gateway access number (DNIS). This macro requires use of the fax detection interactive voice response (IVR) application.
Note Other strings can be passed to mailto in place of $m$ if you modify the fax detection application Tool Command Language (TCL) script or VoiceXML document. For more information, refer to the readme file that came with the TCL script or the Cisco VoiceXML Programmer's Guide.
|
$e$
|
Macro that is replaced by the DNIS, the RDNIS, or a string that represents a valid e-mail address, as specified by the cisco-mailtoaddress variable in the transfer tag of the VoiceXML fax detection document. By default, if the cisco-mailtoaddress variable is not specified in the fax detection document, the DNIS is mapped to $e$.
If $e$ is not specified for the session target mailto command in the MMoIP dial peer, but the cisco-mailtoaddress variable is specified in the transfer tag of the fax detection document, then whatever is specified in the MMoIP dial peer takes precedence; the cisco-mailtoaddress variable is ignored.
Note If a domain name is configured with this command, the VoiceXML document should pass only the username portion of the e-mail address and not the domain. If the domain name is passed from cisco-mailtoaddress, the session target mailto command should specify only $e$.
|
@domain-name
|
(Optional) String that contains the domain name to be associated with the target address, preceded by the at sign (@); for example, @mycompany.com.
|
Command Default
No default behavior or values
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
12.0(4)T
|
This command was modified to support store-and-forward fax.
|
12.1(5)XM1
|
The $m$ keyword was introduced for the fax detection feature on the Cisco AS5300.
|
12.2(2)XA
|
This command was implemented on the Cisco AS5350 and Cisco AS5400.
|
12.2(2)XB
|
The $e$ keyword was introduced for VoiceXML fax detection on the Cisco AS5300.
|
12.2(8)T
|
This command was integrated into Cisco IOS Release 12.2(8)T and was implemented on the following platforms: Cisco 1751, Cisco 2600 series, Cisco 3600 series, Cisco 3725, and Cisco 3745.
|
12.2(11)T
|
This command was implemented on the following platforms: Cisco AS5300, Cisco AS5350, and Cisco AS5400.
|
Usage Guidelines
Use this command to deliver e-mail to one recipient by specifying one e-mail name, or to deliver e-mail to multiple recipients by specifying an e-mail alias as the name argument and having that alias expanded by the mailer.
Use the $m$ macro to include the redirecting dialed number (RDNIS) as part of the e-mail name when using the fax detection IVR application. If $m$ is specified and RDNIS is not present in the call information, the access number of the gateway (the dialed number, or DNIS) is used instead. For example, if the calling party originally dialed 6015551111 to send a fax, and the call was redirected (forwarded on busy or no answer) to 6015552222 (the gateway), the RDNIS is 6015551111, and the DNIS is 6015552222.
Use the $e$ macro to map the cisco-mailtoaddress variable in the VoiceXML fax detection document to the username portion of the e-mail address when sending a fax. If the VoiceXML document does not specify the cisco-mailtoaddress variable in the transfer tag, the application maps the DNIS to the e-mail address username.
Examples
The following example delivers fax-mail to multiple recipients:
session target mailto:marketing-information@mailer.example.com
Assuming that mailer.example.com is running the sendmail application, you can put the following information into its /etc/aliases file:
fax=+14085551212@sj-offramp.example.com
The following example uses the fax detection IVR application. Here, the session target (MMoIP dial peer) command forwards fax calls to an e-mail account that uses the Redirected Dialed Number Identification Service (RDNIS) as part of its address. In this example, the calling party originally dialed 6015551111 to send a fax, and the call was forwarded (on busy or no answer) to 6015552222, which is the incoming number for the gateway being configured. The RDNIS is 6015551111, and the dialed number (DNIS) is 6015552222. When faxes are forwarded from the gateway, the session target in the example is expanded to 6015551111@mail-server.unified-messages.com.
session target mailto:$m$@mail-server.unified-messages.com
The following examples configure a session target for a VoiceXML fax detection application. In this example, the VoiceXML document passes just the username portion of the e-mail address, for example, "johnd":
session target mailto:$e$@cisco.com
In this example, the VoiceXML document passes the complete e-mail address including domain name, for example, "johnd@cisco.com":
session target mailto:$e$
Related Commands
Command
|
Description
|
destination-pattern
|
Specifies either the partial or full E.164 telephone number (depending on your dial plan) used to match the dial peer.
|
dial-peer voice
|
Enters dial peer configuration mode and defines a dial peer.
|
session target (POTS dial peer)
To designate loopback calls from a POTS dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command.
session target loopback:compressed | loopback:uncompressed
no session target
Syntax Description
loopback:compressed
|
All voice data is looped back in compressed mode to the source.
|
loopback:uncompressed
|
All voice data is looped back in uncompressed mode to the source.
|
Command Default
No loopback calls are designated.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series.
|
12.0(3)T
|
This command was implemented on the 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
|
Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400 and Cisco AS5850 is not included in this release.
|
12.2(11)T
|
This command was integrated into Cisco IOS Release 12.2(11)T and is supported on the Cisco AS5200, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
|
Usage Guidelines
Use this command to test the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected.
Examples
The following example loops back the traffic from the dial peer in compressed mode:
session target loopback:compressed
Related Commands
Command
|
Description
|
dial-peer voice
|
Enters dial peer configuration mode and specifies the method of voice-related encapsulation.
|
session target (VoATM dial peer)
To specify a network-specific address for a specified VoATM dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command.
Cisco 3600 Series Routers
session target interface pvc {name | vpi/vci | vci}
no session target
Cisco 7200 Series Routers
session target atm slot/port pvc {word | vpi/vci | vci} cid
no session target
Syntax Description
serial
|
Serial interface for the dial-peer address.
|
atm
|
ATM interface. The only valid number is 0.
|
interface
|
Interface type and interface number on the router.
|
slot/port
|
Slot and port numbers for the dial-peer address.
|
pvc
|
Specific ATM permanent virtual circuit (PVC) for this dial peer.
|
name
|
PVC name.
|
word
|
(Optional) Name that identifies the PVC. The argument can identify the PVC if a word identifier was assigned when the PVC was created.
|
vpi/vci
|
ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC. Values are as follows:
•Cisco 3600 series with Multiport T1/E1 ATM network module with inverse multiplexing over ATM (IMA): vpi range is from 0 to 5; vci range is from 1 to 255.
•OC3 ATM network module: vpi range is from 0 to 15; vci range is from 1 to 1023.
|
vci
|
ATM network virtual channel identifier (VCI) of this PVC.
|
cid
|
ATM network channel identifier (CID) of this PVC. Range is from 8 to 255.
|
Command Default
Command is enabled with no IP address or domain name defined.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(1)MA
|
This command was modified to support VoATM, VoHDLC, and POTS dial peers. The command was implemented on the Cisco MC3810.
|
12.0(3)XG
|
This command was modified to support VoFR dial peers. The command was implemented on the Cisco 2600 series and Cisco 3600 series.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release 12.0(4)T.
|
12.0(7)XK
|
This command was modified to support VoATM and VoIP dial peers. The command was implemented on the Cisco 3600 series and the Cisco MC3810. Support for VoHDLC was removed.
|
12.1(1)XA
|
This command was modified to provide enhanced support for VoATM dial peers.
|
12.1(2)T
|
This command was integrated into Cisco IOS Release 12.1(2)T.
|
12.2(2)T
|
This command was implemented on the Cisco 7200 series.
|
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol that you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.
Use the session target loopback command to test the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected.
This command applies to on-ramp store-and-forward fax functions.
You must enter the session protocol aal2-trunk dial peer configuration command before you can specify a CID for a dial peer for VoATM on the Cisco 7200 series router.
Note This command does not apply to POTS dial peers.
Examples
The following example configures a session target for VoATM. The session target is sent to ATM interface 0 for a PVC with a VCI of 20.
destination-pattern 13102221111
session target atm0 pvc 20
The following example delivers fax-mail to multiple recipients:
session target marketing-information@mailer.example.com
Assuming that mailer.example.com is running sendmail, you can put the following information into its /etc/aliases file:
fax=+14085551212@sj-offramp.example.com
The following example configures a session target for VoATM. The session target is sent to ATM interface 0, and is for a PVC with a VPI/VCI of 1/100.
destination-pattern 13102221111
session target atm1/0 pvc 1/100
Related Commands
Command
|
Description
|
called-number
|
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
|
codec (dial peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
cptone
|
Specifies a regional tone, ring, and cadence setting for an analog voice port.
|
destination-pattern
|
Specifies either the prefix or full E.164 telephone number (depending on the dial plan) to be used for a dial peer.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred selection order of a dial peer within a hunt group.
|
session protocol
|
Establishes a VoFR protocol for calls between local and remote routers via the packet network.
|
session target
|
Configures a network-specific address for a dial peer.
|
session target loopback
|
Tests the voice transmission path of a call.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
session target (VoFR dial peer)
To specify a network-specific address for a specified VoFR dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command.
Cisco 2600 Series and Cisco 3600 Series Routers
session target interface dlci [cid]
no session target
Cisco 7200 Series Routers
session target interface dlci
no session target
Syntax Description
interface
|
Serial interface and interface number (slot number and port number) associated with this dial peer. For the range of valid interface numbers for the selected interface type, enter a ? character after the interface type.
|
dlci
|
Data link connection identifier for this dial peer. Range is from 16 to 1007.
|
cid
|
(Optional) DLCI subchannel to be used for data on FRF.11 calls. A CID must be specified only when the session protocol is frf11-trunk. When the session protocol is cisco-switched, the CID is dynamically allocated. Range is from 4 to 255.
Note By default, CID 4 is used for data; CID 5 is used for call-control. We recommend that you select CID values between 6 and 63 for voice traffic. If the CID is greater than 63, the FRF.11 header contains an extra byte of data.
|
Command Default
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced.
|
11.3(1)MA
|
This command was implemented for VoFR, VoHDLC, and POTS dial peers on the Cisco MC3810.
|
12.0(3)XG
|
This command was implemented for VoFR dial peers on the Cisco 2600 series and Cisco 3600 series. The cid option was added.
|
12.0(4)T
|
This command was integrated into Cisco IOS Release12.0(4)T and implemented for VoFR and POTS dial peers on the Cisco 7200 series.
|
Usage Guidelines
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select. The syntax of this command complies with the simple syntax of mailto: as described in RFC 1738.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point depends on the call origin and the loopback type selected.
For VoFR dial peers, the cid option is not allowed when the cisco-switched option for the session protocol command is used.
Examples
The following example configures serial interface 1/0, DLCI 100 as the session target for Voice over Frame Relay dial peer 200 (an FRF.11 dial peer) using the FRF.11 session protocol:
destination-pattern 13102221111
session protocol frf11-trunk
session target serial 1/0 100 20
The following example delivers fax-mail to multiple recipients:
session target marketing-information@mailer.example.com
Assuming that mailer.example.com is running sendmail, you can put the following information into its /etc/aliases file:
fax=+14085551212@sj-offramp.example.com
Related Commands
Command
|
Description
|
called-number
|
Enables an incoming VoFR call leg to be bridged to the correct POTS call leg.
|
codec (dial peer)
|
Specifies the voice coder rate of speech for a dial peer.
|
cptone
|
Specifies a regional tone, ring, and cadence setting for an analog voice port.
|
destination-pattern
|
Specifies either the prefix or the full E.164 telephone number (depending on the dial plan) to be used for a dial peer.
|
dtmf-relay
|
Enables the DSP to generate FRF.11 Annex A frames for a dial peer.
|
preference
|
Indicates the preferred selection order of a dial peer within a hunt group.
|
session protocol
|
Establishes a VoFR protocol for calls between the local and the remote routers via the packet network.
|
signal-type
|
Sets the signaling type to be used when connecting to a dial peer.
|
session target (VoIP dial peer)
To designate a network-specific address to receive calls from a VoIP dial peer, use the session target command in dial peer configuration mode. To reset to the default, use the no form of this command.
Cisco 1751, Cisco 3725, Cisco 3745, Cisco AS5300
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name |
enum:table-num | loopback:rtp | ras | sip-server}
no session target
Cisco 2600 Series, Cisco 3600 Series, Cisco AS5350, Cisco AS5400, and Cisco AS5850
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name |
enum:table-num | loopback:rtp | ras | settlement provider-number | sip-server}
no session target
Syntax Description
ipv4:destination-address
|
IP address of the dial peer to receive calls.
|
dns:[$s$....] host-name
|
Host device housing the domain name server that resolves the name of the dial peer to receive calls.
Use one of the following macros with this keyword when defining the session target for VoIP peers:
•$s$.—(Optional) Source destination pattern is used as part of the domain name.
•$d$.—(Optional) Destination number is used as part of the domain name.
•$e$.—(Optional) Digits in the called number are reversed and periods are added between the digits of the called number. The resulting string is used as part of the domain name.
•$u$.—(Optional) Unmatched portion of the destination pattern (such as a defined extension number) is used as part of the domain name.
•host-name—String that contains the complete host name to be associated with the target address; for example, serverA.mycompany.com.
|
enum:table-num
|
ENUM search table number. Range is from 1 to 15.
|
loopback:rtp
|
All voice data is looped back to the source.
|
ras
|
Registration, admission, and status (RAS) signaling function protocol is being used, meaning that a gatekeeper is consulted to translate the E.164 address into an IP address.
|
settlement provider-number
|
The settlement server is the target to resolve the terminating gateway address. The argument is as follows:
•provider-number—Provider IP address.
|
sip-server
|
The global Session Initiation Protocol (SIP) server is the destination for calls from this dial peer.
|
Command Default
Enabled, with no IP address or domain name defined.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
11.3(1)T
|
This command was introduced on the Cisco 2600 series and Cisco 3600 series.
|
12.0(3)T
|
This command was implemented on the Cisco AS5300. The ras keyword was added.
|
12.0(4)XJ
|
This command was implemented for store-and-forward fax on the Cisco AS5300.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T. The settlement and sip-server keywords were added.
|
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. Support for on 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. The enum keyword was added.
|
Usage Guidelines
Use this command to specify a network-specific destination for a dial peer to receive calls from the current dial peer. You can select an option to define a network-specific address or domain name as a target, or you can select one of several methods to automatically determine the destination for calls from the current dial peer.
Use the session target dns command with or without the specified macros. Using the optional macros can reduce the number of VoIP dial-peer session targets that you must configure if you have groups of numbers associated with a particular router.
The session target enum command instructs the dial peer to use a table of translation rules to convert the dialed number identification service (DNIS) number into a number in E.164 format. This translated number is sent to a DNS server that contains a collection of URLs. These URLs identify each user as a destination for a call and may represent various access services, such as SIP, H.323, telephone, fax, e-mail, instant messaging, and personal web pages. Before assigning the session target to the dial peer, configure an ENUM match table with the translation rules using the voice enum-match-table command in global configuration mode. The table is identified in session target enum as table-num.
Use the session target loopback command to test the voice transmission path of a call. The loopback point depends on the call origin.
Use the session target ras command to specify that the RAS protocol is being used to determine the IP address of the session target.
In Cisco IOS Release 12.1(1)T the session target command configuration cannot combine the target of RAS with the settle-call command.
If the session target type is settlement when the VoIP dial peers are configured for a settlement server, the provider-number parameter in the session target and settle-call commands should be identical.
Use the session target sip-server command to name the global SIP server interface as the destination for calls from this dial peer. You must first define the SIP server interface by using the sip-server command in SIP UA configuration mode. Then you can enter the session target sip-server option for each dial peer instead of having to enter the entire IP address for the SIP server interface under each dial peer.
Examples
The following example creates a session target using DNS for a host named "voice_router" in the domain cisco.com:
session target dns:voice_router.cisco.com
The following example creates a session target using DNS with the optional $u$. macro. In this example, the destination pattern ends with four periods (.) to allow for any four-digit extension that has the leading numbers 1310555. The optional macro $u$. directs the gateway to use the unmatched portion of the dialed number—in this case, the four-digit extension—to identify a dial peer. As in the preceding example, the domain is "cisco.com."
destination-pattern 1310555....
session target dns:$u$.cisco.com
The following example creates a session target using DNS, with the optional $d$. macro. In this example, the destination pattern has been configured for 13105551111. The optional macro $d$. directs the gateway to use the destination pattern to identify a dial peer in the "cisco.com" domain.
destination-pattern 13105551111
session target dns:$d$.cisco.com
The following example creates a session target using DNS, with the optional $e$. macro. In this example, the destination pattern has been configured for 12345. The optional macro $e$. directs the gateway to do the following: reverse the digits in the destination pattern, add periods between the digits, and use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.
destination-pattern 12345
session target dns:$e$.cisco.com
The following example creates a session target using an ENUM table. It indicates that calls made using dial peer 101 should use the preferential order of rules in enum match table 3.
The following example creates a session target using RAS:
destination-pattern 13105551111
The following example creates a session target using settlement:
session target settlement:0
Related Commands
Command
|
Description
|
destination-pattern
|
Specifies either the prefix or the full E.164 telephone number (depending on the dial plan) to be used for a dial peer.
|
dial-peer voice
|
Enters dial peer configuration mode and specifies the method of voice-related encapsulation.
|
settle-call
|
Specifies that settlement is to be used for the specified dial peer, regardless of session target type.
|
sip-server
|
Defines a network address for the SIP server interface.
|
voice enum-match-table
|
Initiates the ENUM match table definition.
|
session transport
To configure a VoIP dial peer to use TCP or User Datagram Protocol (UDP) as the underlying transport layer protocol for Session Initiation Protocol (SIP) messages, use the session transport command in dial peer configuration mode. To reset to the default (udp keyword), use the no form of this command.
session transport {system | tcp tls | udp}
no session transport {system | tcp tls | udp}
Syntax Description
system
|
The SIP dial peer defers to the voice service VoIP session transport.
|
tcp tls
|
The SIP dial peer uses Transport Layer Security (TLS) over the TCP transport layer protocol.
|
udp
|
The SIP dial peer uses the UDP transport layer protocol. This is the default.
|
Command Default
UDP
Note The transport protocol specified with the transport command must match the one specified with this command.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the 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(11)T
|
This command was integrated into Cisco IOS Release 12.2(11)T.
|
12.4(6)T
|
The tls keyword was added to the command.
|
Usage Guidelines
Use the show sip-ua status command to ensure that the transport protocol that you set using this command matches the protocol set using the transport command. The transport command is used in dial peer configuration mode to specify the SIP transport method, either UDP, TCP, or TLS over TCP.
Examples
The following example shows a VoIP dial peer configured to use TLS over TCP as the underlying transport layer protocol for SIP messages:
session transport tcp tls
The following example shows a VoIP dial peer configured to use UDP as the underlying transport layer protocol for SIP messages:
Related Commands
Command
|
Description
|
show sip-ua status
|
Displays the status of SIP call service on a SIP gateway.
|
transport
|
Configures the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.
|
session transport (H.323 voice-service)
To configure the underlying transport layer protocol for H.323 messages to be used across all VoIP dial peers, use the session transport command in H.323 voice service configuration mode. To reset the default value, use the no form of this command.
session transport {udp | tcp [calls-per-connection value]}
no session transport
Syntax Description
udp
|
Configures the H.323 dial peer to use the UDP transport layer protocol.
|
tcp
|
Configures the H.323 dial peer to use the TCP transport layer protocol. This is the default.
|
calls-per-connection
|
Configures the number of calls multiplexed into a single TCP connection.
|
value
|
The number of calls. The range is from 1 to 9999. The default is 5.
|
Command Default
TCP is the default session transport protocol; the default calls-per-connection value is 5.
Command Modes
H.323 voice service configuration
Command History
Release
|
Modification
|
12.2(1)T
|
This command was introduced for session initiation protocol (SIP) dial peers.
|
12.2(2)XA
|
This command was modified to include support for H323 dial peers and to include the calls-per-connection keyword.
|
12.2(4)T
|
This command was integrated into Cisco IOS Release 12.2(4)T.
|
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.
|
Examples
The following example shows a dial peer configured to use the UDP transport layer protocol.
Router(conf-voi-serv)# h323
Router(conf-serv-h323)# session transport udp
Related Commands
Command
|
Description
|
h323
|
Enables H.323 voice service configuration commands.
|
session transport (SIP)
To configure the underlying transport layer protocol for SIP messages to transport layer security over TCP (TLS over TCP) or User Datagram Protocol (UDP), use the session transport command in SIP configuration mode. To reset the value of this command to the default, use the no form of this command.
session transport {udp | tcp tls}
no session transport {udp | tcp tls}
Syntax Description
udp
|
Configure SIP messages to use the UDP transport layer protocol. This is the default.
|
tcp tls
|
Configure SIP messages to use the TLS over TCP transport layer protocol.
|
Command Default
The default for the command is UDP.
Command Modes
SIP configuration
Command History
Release
|
Modification
|
12.2(2)XB
|
This command was introduced in SIP configuration mode.
|
12.2(2)XB2
|
This command was implemented on the Cisco AS5850 platform.
|
12.2(8)T
|
This command was integrated into Cisco IOS Release 12.2(8)T and support was added for the Cisco 3700 series. Cisco AS5300, Cisco AS5350, Cisco AS5850, and Cisco AS5400 platforms were not supported in this release.
|
12.2(11)T
|
Support was added for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms.
|
12.4(6)T
|
The tls keyword was added to the command.
|
Usage Guidelines
Use the show sip-ua status command to verify that the transport protocol set with the session transport command matches the protocol set using the transport command in SIP user agent configuration mode.
Examples
The following example configures the underlying transport layer protocol for SIP messages to UDP:
The following example configures the underlying transport layer protocol for SIP messages to TLS over TCP:
session transport tcp tls
Related Commands
Command
|
Description
|
show sip-ua status
|
Displays the status of SIP call service on a SIP gateway.
|
transport
|
Configures the SIP gateway for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.
|
session-set
To create a Signlaing System 7 (SS7)-link-to-SS7-session-set association or to associate an SS7 link with an SS7 session set on the Cisco 2600-based Signaling Link Terminal (SLT), enter the session-set command in global configuration mode. To remove the link from its current SS7 session set and to add it to SS7 session set 0 (the default), use the no form of this command.
session-set session-set-id
no session-set
Syntax Description
session-set-id
|
SS7 session ID. Valid values are 0 and 1. Default is 0.
|
Command Default
SS7 session set 0
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.2(15)T
|
This command was introduced on the Cisco 2600-based SLT.
|
Usage Guidelines
On Cisco AS5350 and Cisco AS5400 platforms, the channel-id command is used to create an SS7-link-to-SS7-session-set association on the Cisco SLT. The Cisco 26xx platforms do not support the channel-id command, so channel IDs on the Cisco 26xx-based SLT are implicitly assigned on the basis of the slot location of the WAN interface card (WIC) and the channel group ID used to create the SS7 link.
If this command is omitted, the link is implicitly added to the SS7 session set 0, which is the default.
Examples
The following example shows how the session-set command is used to add the associated SS7 link to an SS7 session set:
The following example shows how the no session-set command is used to remove the link from its current SS7 session set and add it to SS7 session set 0, which is the default:
Related Commands
Command
|
Description
|
channel-id
|
Assigns a session channel ID to a Signaling System 7 (SS7) serial link or assign an SS7 link to an SS7 session set on a Cisco AS5350 or Cisco AS5400.
|
set
To create a fault-tolerant or non-fault-tolerant session set with the client or server option, use the set command in backhaul session-manager configuration mode. To delete the set, use the no form of this command.
set set-name {client | server} {ft | nft}
no set set-name {client | server} {ft | nft}
Syntax Description
set-name
|
Session-set name.
|
client
|
The session set operates as a client. Select this option for signaling backhaul.
|
server
|
The session set operates as a server.
|
ft
|
Fault-tolerant operation. Select fault-tolerant if this session set can contain more than one session group, with each session group connecting the gateway to a different Cisco VSC3000. Fault-tolerance allows the system to operate properly if a session group in the session set fails.
|
nft
|
Non-fault-tolerant operation. Select non-fault-tolerant if this session set contains only one session group (which connects the gateway to a single Cisco VSC3000).
|
Command Default
No default behavior or values
Command Modes
Backhaul session-manager configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced.
|
12.2(4)T
|
This command was implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.
|
12.2(2)XB
|
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 was implemented on the Cisco IAD2420 series. Support for on the Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
|
12.2(11)T
|
This command is supported on the Cisco AS5350, Cisco AS5400 and Cisco AS5850 in this release.
|
Usage Guidelines
Multiple session groups can be associated with a session set. For signaling backhaul, session sets should be configured to operate as clients. A session set cannot be deleted unless all session groups associated with the session set are deleted first.
Examples
The following example sets the client set named "set1" as fault-tolerant:
Router(config-bsm)# set set1 client ft
set http client cache stale
To set the status of all entries in the HTTP client cache to stale, use the set http client cache stale command in global configuration mode. To return to the default, use the no form of this command.
set http client cache stale
no set http client cache stale
Syntax Description
This command has no arguments or keywords.
Command Default
Entries in the HTTP client cache are not marked stale manually.
Command Modes
Global configuration (config)
Command History
Release
|
Modification
|
12.4(18)T
|
This command was introduced.
|
Usage Guidelines
Use this command to force the HTTP client to check with the server to see if an updated version of the file exists when any cached entries are requested by the VoiceXML application. If the router is in nonstreaming mode, a conditional reload is sent to the HTTP server. If the router is in streaming mode, an unconditional reload is sent for the refresh. Regardless of which mode the router is in, the VoiceXML application is guaranteed to receive the most up-to-date file when you use the set http client cache stale command.
The show http client cache command shows a pound sign (#) next to the age of entries that are marked stale manually.
Examples
The following example sets the status of all entries in the HTTP client cache to stale:
Router# set http client cache stale
Related Commands
Command
|
Description
|
show http client cache
|
Displays information about the entries contained in the HTTP client cache.
|
set pstn-cause
To map an incoming PSTN cause code to a Session Initiation Protocol (SIP) error status code, use the set pstn-cause command in SIP UA configuration mode. To reset to the default, use the no form of this command.
set pstn-cause value sip-status value
no set pstn-cause
Syntax Description
pstn-cause value
|
PSTN cause code. Range is from 1 to 127
|
sip-status value
|
SIP status code that is to correspond with the PSTN cause code. Range is from 400 to 699.
|
Command Default
The default mappings defined in the following table are used:
Table 37 Default PSTN Cause Codes Mapped to SIP Events
PSTN Cause Code
|
Description
|
SIP Event
|
1
|
Unallocated number
|
404 Not found
|
2
|
No route to specified transit network
|
404 Not found
|
3
|
No route to destination
|
404 Not found
|
17
|
User busy
|
486 Busy here
|
18
|
No user responding
|
480 Temporarily unavailable
|
19
|
No answer from the user
|
20
|
Subscriber absent
|
21
|
Call rejected
|
403 Forbidden
|
22
|
Number changed
|
410 Gone
|
26
|
Non-selected user clearing
|
404 Not found
|
27
|
Destination out of order
|
404 Not found
|
28
|
Address incomplete
|
484 Address incomplete
|
29
|
Facility rejected
|
501 Not implemented
|
31
|
Normal, unspecified
|
404 Not found
|
34
|
No circuit available
|
503 Service unavailable
|
38
|
Network out of order
|
503 Service unavailable
|
41
|
Temporary failure
|
503 Service unavailable
|
42
|
Switching equipment congestion
|
503 Service unavailable
|
47
|
Resource unavailable
|
503 Service unavailable
|
55
|
Incoming class barred within the Closed User Group (CUG)
|
403 Forbidden
|
57
|
Bearer capability not authorized
|
403 Forbidden
|
58
|
Bearer capability not currently available
|
501 Not implemented
|
65
|
Bearer capability not implemented
|
501 Not implemented
|
79
|
Service or option not implemented
|
501 Not implemented
|
87
|
User not member of the Closed User Group (CUG)
|
503 Service unavailable
|
88
|
Incompatible destination
|
400 Bad request
|
95
|
Invalid message
|
400 Bad request
|
102
|
Recover on Expires timeout
|
408 Request timeout
|
111
|
Protocol error
|
400 Bad request
|
Any code other than those listed above
|
500 Internal server error
|
Command Modes
SIP UA configuration
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. Support for on the Cisco AS5300 Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
|
Usage Guidelines
A PSTN cause code can be mapped only to one SIP status code at a time.
Examples
The following example maps a SIP status code to correspond to a PSTN cause code:
Router(config-sip-ua)# set pstn-cause 111 sip-status 400
Router(config-sip-ua)# exit
Related Commands
Command
|
Description
|
set sip-status
|
Sets an incoming SIP error status code to a PSTN release cause code.
|
set sip-status
To map an incoming Session Initiation Protocol (SIP) error status code to a PSTN cause code, use the set sip-status command in SIP UA configuration mode. To reset to the default, use the no form of this command.
set sip-status value pstn-cause value
no set sip-status
Syntax Description
sip-status value
|
SIP status code. Range is from 400 to 699.
|
pstn-cause value
|
PSTN cause code that is to correspond with the SIP status code. Range is from 1 to 127.
|
Command Default
The default mappings defined in the following table are used:
Table 38 Default SIP Events Mapped to PSTN Cause Codes
SIP Event
|
PSTN Cause Code
|
Description
|
400 Bad request
|
127
|
Interworking, unspecified
|
401 Unauthorized
|
57
|
Bearer capability not authorized
|
402 Payment required
|
21
|
Call rejected
|
403 Forbidden
|
57
|
Bearer capability not authorized
|
404 Not found
|
1
|
Unallocated number
|
405 Method not allowed
|
127
|
Interworking, unspecified
|
406 Not acceptable
|
407 Proxy authentication required
|
21
|
Call rejected
|
408 Request timeout
|
102
|
Recover on Expires timeout
|
409 Conflict
|
41
|
Temporary failure
|
410 Gone
|
1
|
Unallocated number
|
411 Length required
|
127
|
Interworking, unspecified
|
413 Request entity too long
|
414 Request URI (URL) too long
|
415 Unsupported media type
|
79
|
Service or option not available
|
420 Bad extension
|
127
|
Interworking, unspecified
|
480 Temporarily unavailable
|
18
|
No user response
|
481 Call leg does not exist
|
127
|
Interworking, unspecified
|
482 Loop detected
|
483 Too many hops
|
484 Address incomplete
|
28
|
Address incomplete
|
485 Address ambiguous
|
1
|
Unallocated number
|
486 Busy here
|
17
|
User busy
|
487 Request canceled
|
127
|
Interworking, unspecified
|
488 Not acceptable here
|
127
|
Interworking, unspecified
|
500 Internal server error
|
41
|
Temporary failure
|
501 Not implemented
|
79
|
Service or option not implemented
|
502 Bad gateway
|
38
|
Network out of order
|
503 Service unavailable
|
63
|
Service or option unavailable
|
504 Gateway timeout
|
102
|
Recover on Expires timeout
|
505 Version not implemented
|
127
|
Interworking, unspecified
|
580 Precondition failed
|
47
|
Resource unavailable, unspecified
|
600 Busy everywhere
|
17
|
User busy
|
603 Decline
|
21
|
Call rejected
|
604 Does not exist anywhere
|
1
|
Unallocated number
|
606 Not acceptable
|
58
|
Bearer capability not currently available
|
Command Modes
SIP UA configuration
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. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.
|
Usage Guidelines
A SIP status code can be mapped to many PSTN cause codes. For example, 503 can be mapped to 34, 38, and 58.
Examples
The following example maps a PSTN cause code to correspond to a SIP status code:
Router(config-sip-ua)# set sip-status 400 pstn-cause 16
Related Commands
Command
|
Description
|
set pstn-cause
|
Sets an incoming PSTN cause code to a SIP error status code.
|
settle-call
To force a call to be authorized with a settlement server that uses the address resolution method specified in the session target command, use the settle-call command in dial peer configuration mode. To ensure that no authorization is performed by a settlement server, use the no form of this command.
settle-call provider-number
no settle-call provider-number
Syntax Description
provider-number
|
Digit defining the ID of a particular settlement server. The only valid entry is 0.
Note If session target type is settlement, the provider-number argument in the session target and settle-call commands should be identical.
|
Command Default
No default behavior or values.
Command Modes
Dial peer configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
|
Usage Guidelines
With the session target command, a dial peer can determine the address of the terminating gateway through the ipv4, dns, ras, and settlement keywords.
If the session target is not settlement, and the settle-call provider-number argument is set, the gateway resolves address of the terminating gateway using the specified method and then requests the settlement server to authorize that address and create a settlement token for that particular address. If the server cannot authorize the terminating gateway address suggested by the gateway, the call fails.
Do not combine the session target types ras and settle-call. Combination of session target types is not supported.
Examples
The following example sets a call to be authorized with a settlement server that uses the address resolution method specified in the session target:
destination-pattern 1408.......
session target ipv4:172.22.95.14
Related Commands
Command
|
Description
|
session target
|
Specifies a network-specific address for a specified dial peer.
|
settlement
To enter settlement configuration mode and specify the attributes specific to a settlement provider, use the settlement command in global configuration mode. To disable the settlement provider, use the no form of this command.
settlement provider-number
no settlement provider-number
Syntax Description
provider-number
|
Digit that defines a particular settlement server. The only valid entry is 0.
|
Command Default
0
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(4)XH1
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
|
12.1(1)T
|
This command was integrated into Cisco IOS Release 12.1(1)T.
|
Usage Guidelines
The variable provider-number defines a particular settlement provider. For Cisco IOS Release 12.1, only one clearinghouse per system is allowed, and the only valid value for provider-number is 0.
Examples
This example enters settlement configuration mode:
Related Commands
Command
|
Description
|
connection-timeout
|
Configures the length of time for which a connection is maintained after a communication exchange is completed.
|
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.
|
retry-limit
|
Sets the connection retry limit.
|
session-timeout
|
Sets the interval for closing the connection when there is no input or output traffic.
|
show settlement
|
Displays the configuration for all settlement server transactions.
|
shutdown
|
Brings up the settlement provider.
|
type
|
Configures an SAA-RTR operation type.
|
settlement roam-pattern
To configure a pattern that must be matched to determine if a user is roaming, use the settlement roam-pattern command in global configuration mode. To delete a particular pattern, use the no form of this command.
settlement provider-number roam-pattern pattern {roaming | no roaming}
no settlement provider-number roam-pattern pattern {roaming | no roaming}
Syntax Description
provider-number
|
Digit defining the ID of particular settlement server. The only valid entry is 0.
|
pattern
|
User account pattern.
|
roaming | no roaming
|
Whether a user is roaming.
|
Command Default
No default pattern
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.1(1)T
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
|
Usage Guidelines
Multiple roam patterns can be entered on one gateway.
Examples
The following example configures a pattern that determines if a user is roaming:
settlement 0 roam-pattern 1222 roam
settlement 0 roam-pattern 1333 noroam
settlement roam-pattern 1444 roam
settlement roam-pattern 1555 noroam
Related Commands
Command
|
Description
|
roaming (settlement)
|
Enables the roaming capability for a settlement provider.
|
settlement
|
Enters settlement configuration mode.
|
sgcp
To start and allocate resources for the Simple Gateway Control Protocol (SGCP) daemon, use the sgcp command in global configuration mode. To terminate all calls, release all allocated resources, and kill the SGCP daemon, use the no form of this command.
sgcp
no sgcp
Syntax Description
This command has no arguments or keywords.
Command Default
The SGCP daemon is not enabled.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 only and was not generally available.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and Cisco MC3810.
|
Usage Guidelines
When the SGCP daemon is not active, all SGCP messages are ignored.
When you enter the no sgcp command, the SGCP process is removed.
Note After you enter the no sgcp command, you must save the configuration and reboot the router for the disabling of SGCP to take effect.
Examples
The following example enables the SGCP daemon:
The following example disables the SGCP daemon:
Related Commands
Command
|
Description
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp call-agent
To define the IP address of the default Simple Gateway Control Protocol (SGCP) call agent in the router configuration file, use the sgcp call-agent command in global configuration mode. To remove the IP address of the default SGCP call agent from the router configuration, use the no form of this command.
sgcp call-agent ipaddress [:udp port]
no sgcp call-agent ipaddress
Syntax Description
ipaddress
|
IP address or hostname of the call agent.
|
:udp port
|
(Optional) UDP port of the call agent.
|
Command Default
No IP address is configured.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 only and was not generally available.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and Cisco MC3810.
|
Usage Guidelines
This command defines the IP address of the default SGCP call agent to which the router sends an initial RSIP (Restart In Progress) packet when the router boots up. This is used for initial bootup only before the SGCP call agent contacts the router acting as the gateway.
When you enter the no sgcp call-agent command, only the IP address of the default SGCP call agent is removed.
Examples
The following example enables SGCP and specifies the IP address of the call agent:
sgcp call-agent 209.165.200.225
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp graceful-shutdown
To block all new calls and gracefully terminate all existing calls (wait for the caller to end the call), use the sgcp graceful-shutdown command in global configuration mode. To unblock all calls and allow new calls to go through, use the no form of this command.
sgcp graceful-shutdown
no sgcp graceful-shutdown
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 and was not generally available.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810 and Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and Cisco MC3810.
|
Usage Guidelines
Once you issue this command, all requests for new connections (CreateConnection requests) are denied. All existing calls are maintained until users terminate them, or until you enter the no sgcp command. When the last active call is terminated, the SGCP daemon is terminated, and all resources allocated to it are released.
Examples
The following example blocks all new calls and terminates existing calls:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp max-waiting-delay
To set the Simple Gateway Control Protocol (SGCP) maximum waiting delay to prevent restart avalanches, use the sgcp max-waiting-delay command in global configuration mode. To reset to the default, use the no form of this command.
sgcp max-waiting-delay delay
no sgcp max-waiting-delay delay
Syntax Description
delay
|
Maximum waiting delay (MWD), in milliseconds. Range is from 0 to 600000. Default is 3000.
|
Command Default
3,000 ms
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300, and was not generally available.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
|
Examples
The following example sets the maximum wait delay value to 40 ms:
sgcp max-waiting-delay 40
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp modem passthru
To enable Simple Gateway Control Protocol (SGCP) modem or fax pass-through, use the sgcp modem passthru command in global configuration mode. To disable SGCP modem or fax pass-through, use the no form of this command.
sgcp modem passthru {ca | cisco | nse}
no sgcp modem passthru {ca | cisco | nse}
Syntax Description
ca
|
Call-agent-controlled modem upspeed-method violation message.
|
cisco
|
Cisco-proprietary upspeed method based on the protocol.
|
nse
|
NSE-based modem upspeed method.
|
Command Default
SGCP modem or fax pass-through is disabled by default.
Command Modes
Global configuration.
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 and the Cisco 3600 series (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
|
Usage Guidelines
You can use this command for fax pass-through because the answer tone can come from either modem or fax transmissions. The upspeed method is the method used to dynamically change the codec type and speed to meet network conditions.
If you use the nse option, you must also configure the sgcp tse payload command.
Examples
The following example configures SGCP modem pass-through using the call-agent upspeed method:
The following example configures SGCP modem pass-through using the proprietary Cisco upspeed method:
sgcp modem passthru cisco
The following example configures SGCP modem pass-through using the NSE-based modem upspeed:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp quarantine-buffer disable
To disable the Simple Gateway Control Protocol (SGCP) quarantine buffer, use the sgcp quarantine-buffer disable command in global configuration mode. To reenable the SGCP quarantine buffer, use the no form of this command.
sgcp quarantine-buffer disable
no sgcp quarantine-buffer disable
Syntax Description
This command has no arguments or keywords.
Command Default
The SGCP quarantine buffer is enabled.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was on the Cisco 3600 series and the Cisco MC3810.
|
Usage Guidelines
The SGCP quarantine buffer is the mechanism for buffering the SGCP events between two notification-request (RQNT) messages.
Examples
The following example disables the SGCP quarantine buffer:
sgcp quarantine-buffer disable
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp request retries
To specify the number of times to retry sending notify and delete messages to the Simple Gateway Control Protocol (SGCP) call agent, use the sgcp request retries command in global configuration mode. To reset to the default, use the no form of this command.
sgcp request retries count
no sgcp request retries
Syntax Description
count
|
Number of times that a notify and delete message is retransmitted to the SGCP call agent before it is dropped. Range is from 1 to 100. Default is 3.
|
Command Default
3 times
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 and was not generally available.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
|
Usage Guidelines
The actual retry count may be different from the value you enter for this command. The retry count is also limited by the call agent. If there is no response from the call agent after 30 seconds, the gateway does not retry anymore, even though the number set using the sgcp request retries command has not been reached.
The router stops sending retries after 30 seconds, regardless of the setting for this command.
Examples
The following example configures the system to send the sgcp command 10 times before dropping the request:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp request timeout
To specify how long the system should wait for a response to a request, use the sgcp request timeout command in global configuration mode. To reset to the default, use the no form of this command.
sgcp request timeout timeout
no sgcp request timeout
Syntax Description
timeout
|
Time to wait for a response to a request, in milliseconds. Range is from 1 to 10000. Default is 500.
|
Command Default
500 ms
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 and was not generally available.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
|
Usage Guidelines
This command is used for "notify" and "delete" messages, which are sent to the SGCP call agent.
Examples
The following example configures the system to wait 40 ms for a reply to a request:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp restart
To trigger the router to send a Restart in Progress (RSIP) message to the Simple Gateway Control Protocol (SGCP) call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller, use the sgcp restart command in global configuration mode. To reset to the default, use the no form of this command.
sgcp restart {delay delay | notify}
no sgcp restart {delay delay | notify}
Syntax Description
delay delay
|
Restart delay, in milliseconds. Range is from 0 to 600. Default is 0.
|
notify
|
Restarts notification upon the SGCP/digital interface state transition.
|
Command Default
0 ms
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 and the Cisco 3600 series (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
|
Usage Guidelines
Use this command to send RSIP messages from the router to the SGCP call agent. RSIP messages are used to synchronize the router and the call agent. RSIP messages are also sent when the sgcp command is entered to enable the SGCP daemon.
You must enter the notify option to enable RSIP messages to be sent.
Examples
The following example configures the system to wait 40 ms before restarting SGCP:
The following example configures the system to send an RSIP notification to the SGCP call agent when the T1 controller state changes:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp retransmit timer
To configure the Simple Gateway Control Protocol (SGCP) retransmission timer to use a random algorithm, use the sgcp retransmit timer command in global configuration mode. To reset to the default, use the no form of this command.
sgcp retransmit timer {random}
no sgcp retransmit timer {random}
Syntax Description
random
|
SGCP retransmission timer uses a random algorithm.
|
Command Default
The SGCP retransmission timer does not use a random algorithm.
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco 3600 series and the Cisco MC3810 in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
|
Usage Guidelines
Use this command to enable the random algorithm component of the retransmission timer. For example, if the retransmission timer is set to 200 ms, the first retransmission timer is 200 ms, but the second retransmission timer picks up a timer value randomly between either 200 or 400. The third retransmission timer picks up a timer value randomly of 200, 400, or 800 as shown below:
•First retransmission timer: 200
•Second retransmission timer: 200 or 400
•Third retransmission timer: 200, 400, or 800
•Fourth retransmission timer: 200, 400, 800, or 1600
•Fifth retransmission timer: 200, 400, 800, 1600, or 3200 and so on.
After 30 seconds, the retransmission timer no longer retries.
Examples
The following example sets the retransmission timer to use a random algorithm:
sgcp retransmit timer random
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp timer
To configure how the gateway detects the Real-Time Transport Protocol (RTP) stream lost, use the sgcp timer command in global configuration mode. To reset to the default, use the no form of this command.
sgcp timer {receive-rtcp timer | rtp-nse timer}
no sgcp timer {receive-rtcp timer | rtp-nse timer}
Syntax Description
receive-rtcp timer
|
RTP Control Protocol (RTCP) transmission interval, in milliseconds. Range is from 1 to 100. Default is 5.
|
rtp-nse timer
|
RTP named signaling event (NSE) timeout, in milliseconds. Range is from 100 to 3000. Default is 200.
|
Command Default
receive-rtcp: 5 ms
rtp-nse: 200 ms
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(5)T
|
This command was introduced in a private release on the Cisco AS5300 and was not generally available.
|
12.0(7)XK
|
This command was implemented on the Cisco MC3810 and the Cisco 3600 series (except for the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
|
Usage Guidelines
The RTP NSE timer is used for proxy ringing (the ringback tone is provided at the originating gateway).
Examples
The following example sets the RTPCP transmission interval to 100 ms:
sgcp timer receive-rtcp 100
The following example sets the NSE timeout to 1000 ms:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.
|
sgcp tse payload
|
Enables Inband TSE for fax/modem operation.
|
sgcp tse payload
To enable Inband Telephony Signaling Events (TSE) for fax and modem operation, use the sgcp tse payload command in global configuration mode. To reset to the default, use the no form of this command.
sgcp tse payload type
no sgcp tse payload type
Syntax Description
type
|
TSE payload type. Range is from 96 to 119. Default is 0, meaning that the command is disabled.
|
Command Default
0 (disabled)
Command Modes
Global configuration
Command History
Release
|
Modification
|
12.0(7)XK
|
This command was introduced on the Cisco MC3810 and the Cisco 3600 series (except the Cisco 3620) in a private release that was not generally available.
|
12.1(2)T
|
This command was implemented on the Cisco 3600 series and the Cisco MC3810.
|
Usage Guidelines
Because this command is disabled by default, you must specify a TSE payload type.
If you set the sgcp modem passthru command to the nse value, then you must configure this command.
Examples
The following example sets Simple Gateway Control Protocol (SGCP) modem pass-through using the NSE-based modem upspeed and the Inband Telephony Signaling Events payload value set to 110:
Related Commands
Command
|
Description
|
sgcp
|
Starts and allocates resources for the SGCP daemon.
|
sgcp call-agent
|
Defines the IP address of the default SGCP call agent.
|
sgcp graceful-shutdown
|
Gracefully terminates all SGCP activity.
|
sgcp max-waiting-delay
|
Sets the SGCP maximum waiting delay to prevent restart avalanches.
|
sgcp modem passthru
|
Enables SGCP modem or fax pass-through.
|
sgcp quarantine-buffer disable
|
Disables the SGCP quarantine buffer.
|
sgcp request retries
|
Specifies the number of times to retry sending "notify" and "delete" messages to the SGCP call agent.
|
sgcp request timeout
|
Specifies how long the system should wait for a response to a request.
|
sgcp restart
|
Triggers the router to send an RSIP message to the SGCP call agent indicating that the T1 controller is up or down so that the call agent can synchronize with the T1 controller.
|
sgcp retransmit timer
|
Configures the SGCP retransmission timer to use a random algorithm method.up or down so that the call agent can synchronize
|
sgcp timer
|
Configures how the gateway detects the RTP stream host.
|