- A
- B
- cac master through call application stats
- call application voice through call denial
- call fallback through called-number (dial peer)
- caller-id (dial peer) through ccm-manager switchover-to-backup
- ccs connect (controller) through clear vsp statistics
- clid through credentials (sip-ua)
- default (auto-config application) through direct-inward-dial
- disable-early-media through dualtone
- E
- F
- G
- H
- icpif through irq global-request
- isdn bind-l3 through ixi transport http
- K
- L
- map q850-cause through mgcp package-capability
- mgcp persistent through mmoip aaa send-id secondary
- mode (ATM/T1/E1 controller) through mwi-server
- N
- O
- package through pattern
- periodic-report interval through proxy h323
- Q
- R
- sccp through service-type call-check
- session through sgcp tse payload
- show aal2 profile through show call filter match-list
- show call history fax through show debug condition
- show dial-peer through show gatekeeper zone prefix
- show gateway through show modem relay statistics
- show mrcp client session active through show sip dhcp
- show sip service through show trunk hdlc
- show vdev through show voice statistics memory-usage
- show voice trace through shutdown (voice-port)
- signal through srv version
- ss7 mtp2-variant through switchover method
- target carrier-id through timeout tsmax
- timeouts call-disconnect through timing clear-wait
- timing delay-duration through type (voice)
- U
- vad (dial peer) through voice-class sip encap clear-channel
- voice-class sip error-code-override through vxml version 2.0
- W
- Z
- session
- session group
- session protocol (dial peer)
- session protocol (Voice over Frame Relay)
- session protocol aal2
- session protocol multicast
- session refresh
- 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
Command Default
No default behavior or values
Command Modes
Backhaul session manager configuration
Command History
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
Command Default
No default behavior or values.
Command Modes
Backhaul session-manager configuration
Command History
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
Command Default
No default behaviors or values
Command Modes
Dial peer configuration
Command History
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:
dial-peer voice 10 voatm
session protocol aal2-trunk
The following example shows that Cisco session protocol has been configured as the session protocol:
dial-peer voice 20 voip
session protocol cisco
The following example shows that a VoIP dial peer for SIP has been configured as the session protocol for VoIP call signaling:
dial-peer voice 102 voip
session protocol sipv2
Related Commands
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
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:
dial-peer voice 200 vofr
session protocol frf11-trunk
called-number 5552150
Related Commands
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
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:
voice service voatm
session protocol aal2
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
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:
dial-peer voice 111 voip
destination-pattern 111
session protocol multicast
session target ipv4:237.111.0.111:22222
ip precedence 5
codec g711ulaw
Related Commands
|
|
|
|---|---|
session target ipv4 |
Assigns the session target for voice-multicasting dial peers. |
session refresh
To enable SIP session refresh globally, use the session refresh command in SIP configuration mode. To disable the session refresh, use the no form of this command.
session refresh
no session refresh
Syntax Description
This command has no arguments or keywords.
Command Default
No session refresh
Command Modes
SIP configuration (conf-serv-sip)
Command History
|
|
|
|---|---|
15.1(2)T |
This command was introduced. |
Usage Guidelines
Use the SIP session refresh command to send the session refresh request.
Examples
The following example sets the session refresh under SIP configuration mode:
Router(conf-serv-sip)# Session refresh
Related Commands
|
|
|
|---|---|
voice-class sip session refresh |
Enables session refresh at dial-peer level. |
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
Command Default
No default behavior or values
Command Modes
Application configuration
Command History
|
|
|
|---|---|
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:
application
session start my_instance2 demo
Related Commands
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 |
$e$ |
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 |
@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
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:
dial-peer voice 10 mmoip
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:
marketing-information:
john@example.com,
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.
dial-peer voice 4 mmoip
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":
dial-peer voice 4 mmoip
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":
dial-peer voice 5 mmoip
session target mailto:$e$
Related Commands
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
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:
dial-peer voice 10 pots
session target loopback:compressed
Related Commands
|
|
|
|---|---|
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
Command Default
Command is enabled with no IP address or domain name defined.
Command Modes
Dial peer configuration
Command History
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.
dial-peer voice 12 voatm
destination-pattern 13102221111
session target atm0 pvc 20
The following example delivers fax-mail to multiple recipients:
dial-peer voice 10 mmoip
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:
marketing-information:
john@example.com,
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.
dial-peer voice 12 voatm
destination-pattern 13102221111
session target atm1/0 pvc 1/100
Related Commands
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
Command Default
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial peer configuration
Command History
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:
dial-peer voice 200 vofr
destination-pattern 13102221111
called-number 5552150
session protocol frf11-trunk
session target serial 1/0 100 20
The following example delivers fax-mail to multiple recipients:
dial-peer voice 10 mmoip
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:
marketing-information:
john@example.com,
fax=+14085551212@sj-offramp.example.com
Related Commands
session target (VoIP dial peer)
To designate a network-specific address to receive calls from a VoIP or VoIPv6 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, and Cisco AS5300
session target {dhcp | ipv4:destination-address | ipv6:[destination-address] | dns:[$s$. | $d$. | $e$. | $u$.] hostname | enum:table-num | loopback:rtp | ras | sip-server | registrar} [:port]
no session target
Cisco 2600 Series, Cisco 3600 Series, Cisco AS5350, Cisco AS5400, and Cisco AS5850
session target {dhcp | ipv4:destination-address | ipv6:[destination-address] | dns:[$s$. | $d$. | $e$. | $u$.] hostname | enum:table-num | loopback:rtp | ras | settlement provider-number | sip-server | registrar} [:port]
no session target
Syntax Description
Command Default
No IP address or domain name is defined.
Command Modes
Dial peer configuration (config-dial-peer)
Command History
Usage Guidelines
Use the session target 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 the session target enum command with the table-num argument.
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 dhcp command to specify that the session target host is obtained via DHCP. The dhcp option can be made available only if the SIP is being used as the session protocol. To enable SIP, use the session protocol (dial peer) command.
In Cisco IOS Release 12.1(1)T the session target command configuration cannot combine the target of RAS with the settle-call command.
For the session target settlement provider-number command, when the VoIP dial peers are configured for a settlement server, the provider-number argument 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 the dial peer. You must first define the SIP server interface by using the sip-server command in SIP user-agent (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.
After the SIP endpoints are registered with the SIP registrar in the hosted unified communications (UC), you can use the session target registrar command to route the call automatically to the registrar end point. You must configure the session target command on a dial pointing towards the end point.
Examples
The following example shows how to create a session target using DNS for a host named "voicerouter" in the domain example.com:
dial-peer voice 10 voip
session target dns:voicerouter.example.com
The following example shows how to create 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 number 1310555. The optional $u$. macro directs the gateway to use the unmatched portion of the dialed number—in this case, the four-digit extension—to identify a dial peer. The domain is "example.com."
dial-peer voice 10 voip
destination-pattern 1310555....
session target dns:$u$.example.com
The following example shows how to create a session target using DNS, with the optional $d$. macro. In this example, the destination pattern has been configured to 13105551111. The optional macro $d$. directs the gateway to use the destination pattern to identify a dial peer in the "example.com" domain.
dial-peer voice 10 voip
destination-pattern 13105551111
session target dns:$d$.example.com
The following example shows how to create a session target using DNS, with the optional $e$. macro. In this example, the destination pattern has been configured to 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 "example.com" domain.
dial-peer voice 10 voip
destination-pattern 12345
session target dns:$e$.example.com
The following example shows how to create a session target using an ENUM match table. It indicates that calls made using dial peer 101 should use the preferential order of rules in enum match table 3:
dial-peer voice 101 voip
session target enum:3
The following example shows how to create a session target using DHCP:
dial-peer voice 1 voip
session protocol sipv2
voice-class sip outbound-proxy dhcp
session target dhcp
The following example shows how to create a session target using RAS:
dial-peer voice 11 voip
destination-pattern 13105551111
session target ras
The following example shows how to create a session target using settlement:
dial-peer voice 24 voip
session target settlement:0
The following example shows how to create a session target using IPv6 for a host at 2001:10:10:10:10:10:10:230a:5090:
dial-peer voice 4 voip
destination-pattern 5000110011
session protocol sipv2
session target ipv6:[2001:0DB8:10:10:10:10:10:230a]:5090
codec g711ulaw
The following example shows how to configure Cisco Unified Border Element (UBE) to route a call to the registering end point:
dial-peer voice 4 voip
session target registrar
Related Commands
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
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
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:
dial-peer voice 102 voip
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:
dial-peer voice 102 voip
session transport udp
Related Commands
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
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
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
|
|
|
|---|---|
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
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:
voice service voip
sip
session transport udp
The following example configures the underlying transport layer protocol for SIP messages to TLS over TCP:
voice service voip
sip
session transport tcp tls
Related Commands
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
|
|
|
|---|---|
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:
session-set 1
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:
no session-set
Related Commands
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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
Command Default
No default behavior or values
Command Modes
Backhaul session-manager configuration
Command History
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.
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
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12.4(15)T |
This command was introduced. |
12.4(20)T |
This command was integrated into Cisco IOS Release 12.4(20)T. |
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
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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:
Command Modes
SIP UA configuration
Command History
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
Router(config-sip-ua)# set pstn-cause 111 sip-status 400
Router(config-sip-ua)# exit
Related Commands
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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:
Command Modes
SIP UA configuration
Command History
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
Router(config-sip-ua)# set sip-status 400 pstn-cause 16
Related Commands
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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
Command Default
No default behavior or values.
Command Modes
Dial peer configuration
Command History
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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:
dial-peer voice 10 voip
destination-pattern 1408.......
session target ipv4:172.22.95.14
settle-call 0
Related Commands
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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
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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:
settlement 0
Related Commands
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 | noroaming}
no settlement provider-number roam-pattern pattern {roaming | noroaming}
Syntax Description
Command Default
No default pattern is configured.
Command Modes
Global configuration (config)
Command History
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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 shows how to configure a pattern that determines if a user is roaming:
settlement 0 roam-pattern 1222 roaming
settlement 0 roam-pattern 1333 noroaming
settlement 0 roam-pattern 1444 roaming
settlement 0 roam-pattern 1555 noroaming
Related Commands
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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
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:
sgcp
The following example disables the SGCP daemon:
no sgcp
Related Commands
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
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
sgcp call-agent 209.165.200.225
Related Commands
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
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:
sgcp graceful-shutdown
Related Commands
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
Examples
The following example sets the maximum wait delay value to 40 ms:
sgcp max-waiting-delay 40
Related Commands
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
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:
sgcp modem passthru ca
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:
sgcp modem passthru nse
sgcp tse payload 110
Related Commands
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
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
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
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:
sgcp request retries 10
Related Commands
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
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:
sgcp request timeout 40
Related Commands
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
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:
sgcp restart delay 40
The following example configures the system to send an RSIP notification to the SGCP call agent when the T1 controller state changes:
sgcp restart notify
Related Commands
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
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
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
Command Default
receive-rtcp: 5 ms
rtp-nse: 200 ms
Command Modes
Global configuration
Command History
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:
sgcp timer rtp-nse 1000
Related Commands
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
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:
sgcp modem passthru nse
sgcp tse payload 110
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