This chapter covers session configuration objects, A through M, alphabetically. For session configuration objects N through Z, see "Configuring Session Configuration Objects N Through Z." The session configuration objects define the way in which the ME handles SIP-based signaling and media traffic. The session configuration that is applied to an active call through the ME depends on configuration of other aspects of the system.
There are several places in the configuration hierarchy through which you can access the session configuration objects. The path to these object defines in which cases the ME uses that configuration. Locations for session configuration are defined in the following table.
Table 39-1 Session Configuration Locations
| Path | Defines... |
|---|---|
|
vsp > default-session-config |
The session configuration settings to apply to those SIP calls for which there are no configured policies. See Configuring Default Session Configuration Objects, for more information. |
|
vsp > policies > session-policies > policy > rule |
The session configuration settings to apply to SIP calls for which a configured policy exists. See Configuring Policy Objects, for more information. |
|
vsp > dial-plan > dial-prefix vsp > dial-plan > route vsp > dial-plan > source-route |
The session configuration settings to apply to calls based on the dial prefix or domain suffix. See Configuring Dial Plan Objects, for more information. |
|
vsp > calling-group > route vsp > calling-group > source-route |
The session configuration settings to apply to calling-group member calls based on the dial prefix or domain suffix. See Configuring Calling Group Objects, for more information. |
|
vsp > session-config-pool > entry |
A saved session configuration that can be referenced within one or more dial plans. See Configuring Session Configuration Pool Objects, for more information. |
Configures 3rd Generation Partnership Project (3GPP) systems.
config vsp default-session-config 3GPP config vsp policies session-policies policy name rule name session-config 3GPP config vsp dial-plan dial-prefix entryName session-config 3GPP config vsp dial-plan route name session-config 3GPP config vsp dial-plan source-route name session-config 3GPP config vsp session-config-pool entry name 3GPP
Sets the target destination for call detail records. By selecting a target, you are configuring the ME to provide call logging of this SIP session. The records are then sent to the server, database, or file specified in this object. Note that you must configure the destination devices first, and then reference them here.
config vsp default-session-config accounting config vsp policies session-policies policy name rule name session-config accounting config vsp dial-plan dial-prefix entryName session-config accounting config vsp dial-plan route name session-config accounting config vsp dial-plan source-route name session-config accounting config vsp session-config-pool entry name accounting
target: Sets the destination for the accounting records (SIP call detail records) created for this SIP session. When you set the target, you must specify a previously configured object, dependent on the target type:
Default: There is no default setting
Values: radius <radiusGroup>: Logs the session to the group specified in the radius-group object.
diameter <diameterGroup>: Logs the session to the group specified in the diameter-group object.
database <databaseGroup>: Logs the session to an internal or external database group, as specified in the database object diameter-group subobject.
syslog <syslogGroup>: Logs the session to an external syslog server group, as specified in the syslog object diameter-group subobject.
file-system <path>: Writes the session to a file on the ME device, as specified in the file-system > path subobject.
Example: set target syslog ”vsp accounting syslog group Boston”
header: Specifies a string that is written to each accounting record. Use this, for example, to track for later analysis a header that certain user agents output in their INVITE. This header can be seen in the Arbitrary Header field of the Call Record displayed through the ME Management System Call Logs tab.
Default: There is no default setting
Example: set header UA1
accept-mode: Specifies the activity that initiates the connect time in the accounting record. When disabled, the default, the connect time is recorded when the system transmits an ACK. When enabled, the connect time is recorded when the system receives a 200 OK message.
Default: disabled
Values: enabled | disabled
Example: set accept-mode enabled
disconnect-time-upon-receipt-of-bye: Enables or disables logging of the call disconnect time entry in the call detail record when the call session terminates with a BYE request. If set to disabled (the default), the disconnect time is recorded with the 200OK that follows the BYE request.
Default: disabled
Values: enabled | disabled
Example: set disconnect-time-upon-receipt-of-bye enabled
use-short-gateway-names: Specifies how the ME handles particular gateway fields in the CDR. When enabled (the default), the system populates the OrigGW and TermGW fields in the CDR with the server-pool > server-pool-admission-control name string for the originating and terminating SIP server gateway for a call, if known.
If set to disabled, the ME populates the OrigGW and TermGW fields in the CDR with the server name (as configured) followed by a ”- ”and then the server-pool > server-pool-admission-control name.
Default: enabled
Values: enabled | disabled
Example: set use-short-gateway-names disabled
reported-failed-calls: Secondary property. Specifies whether to send out accounting records for calls that did not connect. When disabled, the records are not sent. When enabled, a record of the call is sent to the target configured in your session configuration.
Default: disabled
Values: enabled | disabled
Example: set report-failed-calls enabled
Adds a custom data field to the accounting record. Use this object to define the content of the field. The ME supports several predefined selections for use with the entry property. (You can also add text string values.) To enter a predefined value, precede the letter with slash (and quotation marks in the CLI). For example, ”\b” to add a box identifier. The following table shows the predefined selections for use with this object.
Table 39-2 Predefined Selections For This Object
| Use | For... |
|---|---|
|
\b |
Box identifier |
|
\d |
Digest Realm |
|
\s |
Source LNP |
|
\e |
Destination LNP |
|
\v |
Diversion Header |
|
\p |
P-Charging-Vector Header |
|
\c |
Cluster Name |
|
\r |
RADIUS Caller ID |
|
\z |
Call Connected Boolean |
|
\y |
File play - scan time |
|
\x |
File play - file duration |
|
\w |
File play - playback duration |
|
\u |
Disconnect reason text |
|
\t |
Disconnect reason code |
|
\q |
Post Dial Digits Info |
|
\j |
Source Jitter |
|
\m |
Source Max Jitter |
|
\i |
Destination Jitter |
|
\k |
Destination Max Jitter |
config vsp default-session-config accounting-data config vsp policies session-policies policy name rule name session-config accounting-data config vsp dial-plan dial-prefix entryName session-config accounting-data config vsp dial-plan route name session-config accounting-data config vsp dial-plan source-route name session-config accounting-data config vsp session-config-pool entry name accounting-data
entry: Specifies the content of the field added to the accounting record, in the format tag=value. Use this property, for example, to add a source based on calls matching this session configuration. Use the predefined elements (see the Purpose) to extract data from the call SIP headers.
Default: There is no default setting
Example: set entry cluster-name ”\c”
post-process-expression: Configures a regular expression, and replacement text, to run against the TO and FROM fields of the CDR. Use post processing to, for example, remove unwanted commas that may appear as the result of data that was imported from a CSV file. The following example turns a comma into a dash.
For more information regarding configuring regular expressions and replacement strings, see Using Regular Expressions.
Default: There is no default setting
Example: set post-process-expression (.*0, (.*) ”\1 - 1\2”
custom-data-grouping: The characters used to associate custom data tags and values.
Default: =
Example: set custom-data-grouping ^ custom-data-delimiter: The characters used to separate group data entries.
Default: ;
Example: set custom-data-delimiter ^
named-variable-entry: Specifies the content of the field added to the accounting record in the format, tag=value.
Default: There is no default setting
Example: set named-variable-entry \s
The added-body configuration object allows you to add one or more SIP message body parts.
config vsp default-session-config header-settings added-body config vsp policies session-policies policy name rule name session-config header-settings added-body config vsp dial-plan dial-prefix entryName session-config header-settings added-body config vsp dial-plan route name session-config header-settings added-body config vsp dial-plan source-route name session-config header-settings added-body config vsp session-config-pool entry name header-settings added-body config vsp policies session-policies policy default rule session-config inbound-header-settings added-body
admin: Indicates whether or not the ME is adding the configured SIP message body part.
Default: enabled
Values: enabled | disabled
Example: set admin disabled
content-type: Specifies the type of message body part to add.
Example: set content-type body
other-headers: Specify other headers to include with this added body party.
Default: There is no default setting
Example: set other-headers head1 info
content: Specify the body part content being added.
Default: There is no default setting
Example: set content message body
apply-to-methods: Specifies the message type to which the ME applies message body part changes.
Default: There is no default setting
Example: set apply-to-methods INVITE
apply-to-responses: Specifies whether to apply message body part changes to SIP requests or both requests and responses.
Default: No
Values: No: Apply to requests only.
Yes <response-code>: Apply to responses with the specified response-code.
Both: Apply to both responses and requests.
Example: set apply-to-responses both
apply-to-dialog: Specifies whether to apply message body part changes to a specific dialog or not.
Default: both
Values: both: Apply to both inbound and outbound dialogs
inbound: Apply to inbound dialog only.
outbound: Apply to outbound dialog only.
Example: set apply-to-dialog inbound
cseq: Secondary property. Sets a mechanism that further filters which SIP messages have the body part modifications applied.
Default: 0
Example: set cseq 5
create-on-failed-match: Secondary property. Specifies whether the ME should create a header even if the expression is not a complete match.
Default: true
Values: true | false
Example: set create-on-failed-match false
Modifies the SIP message body in calls matching this session configuration. The ME searches the SIP message body for the specified string and replaces the string as required.
config vsp default-session-config header-settings altered-body number config vsp policies session-policies policy name rule name session-config header-settings altered-body number config vsp dial-plan dial-prefix entryName session-config header-settings altered-body number config vsp dial-plan route name session-config header-settings altered-body number config vsp dial-plan source-route name session-config header-settings altered-body number config vsp session-config-pool entry name header-settings altered-body number
admin: Enables or disables this configuration entry.
Default: There is no default setting
Example: set admin enabled
altered-body: Specifies the text to match on in the message body and the replacement text for the matched string.
For more information regarding configuring regular expressions and replacement strings, see Using Regular Expressions.
Default: There is no default setting
Example: set altered-body "(?ms)(.*)<address uri=\""sip:(.*)@(.*):5060;(.*)" "\1<address uri=" "sip:\2@10.1.1.1:5060;\4"
apply-to-methods: Specifies the message type to which the system applies message body changes. The system then changes the SIP message body (if a match occurs) in all messages of that type.
When you modify this value, the system overwrites the current setting with only the message types you specify. For example, if set to the default and you enter CONTACT, the system only alters CONTACT messages. Enter multiple types separated by a plus sign (+) with no spaces.
Default: INVITE
Example: set apply-to-methods INVITE+CONTACT apply-to-responses: Specifies whether to apply message body changes to SIP requests or requests and responses. Set to no to apply changes only to requests. Set to yes to apply to responses as well. If yes, you must set the response code to which it applies. Create additional altered-body profiles to change multiple response types.
Default: no Values: no | yes responseCode
Example: set apply-to-responses yes 200
cseq: Secondary property. Sets a mechanism to further filter which SIP messages have the header expression modifications applied. If cseq is set to zero (the default), the ME applies the changes to all SIP messages. If set to any other value, the system only applies the changes to SIP messages having a CSEQ field that matches that value.
Default: 0
Example: set cseq 100
remove-body: Secondary property. When this property is set to true, the ME removes the SIP message body from the matching of SIP messages. This includes the ”Content-Type” and other related headers.
Default: false
Values: true | false
Example: set remove-body true
apply-to-dialog: Allows you to configure where to apply these options for a session.
Default: both
Values: inbound: Apply to inbound dialog only.
outbound: Apply to outbound dialog only.
both: Apply to both inbound and outbound dialogs.
Example: set apply-to-dialog inbound
Modifies or creates header values in calls matching this session configuration. Both this and the reg-ex-header objects provide this functionality. Use this object, which is simpler, when possible. Use the reg-ex-header object for complex modification, for example, when multiple levels of change are required. Note that you can create multiple header-altering configurations. They are processed by the system in the order that they appear in the configuration.
config vsp default-session-config header-settings altered-header number config vsp policies session-policies policy name rule name session-config header-settings altered-header number config vsp dial-plan dial-prefix entryName session-config header-settings altered-header number config vsp dial-plan route name session-config header-settings altered-header number config vsp dial-plan source-route name session-config header-settings altered-header number config vsp session-config-pool entry name header-settings altered-header number
admin: Enables or disables this configuration entry.
Default: There is no default setting
Values: enabled | disabled
Example: set admin enabled
source-header: Specifies the URI from which the system initially derives the data that is to be written to the destination header.
Default: There is no default setting
Example: set source-header to
source-field: Specifies the portion of the URI that the system writes to the destination. Select either user, host, selection, or value. With the user and host options, the system writes the entire field to the destination. If you choose selection, specify the value within the URI to match on and the replacement text to write to the destination. Or, select value to write a specific string to the destination. In this case, the source-header field is ignored.
For more information regarding configuring regular expressions and replacement strings, see Using Regular Expressions.
Default: There is no default setting Values: user | host | selection regEx replacement | value replacement
Example: set source-field user
destination-header: Specifies the header to be created or modified by the properties set in this object. The system modifies this URI with the data from the source. If the header does not exist in the message, the system creates it.
Default: There is no default setting
Example: set destination-header request
destination-field: Specifies the field in the specified destination URI to overwrite. Select either user, host , or display. To overwrite the entire selected destination URI, select full.
Default: There is no default setting
Values: user | host | display | full
Example: set destination-field full
apply-to-methods: Specifies the message type to which the system applies header value changes. The system then changes the specified URI according to the settings of the header and destination properties of this object.
When you modify this value, the system overwrites the current setting with only the message types you specify. For example, if set to the default and you enter INVITE, the system only authenticates INVITE messages. Enter multiple types separated by a plus sign (+) with no spaces.
Default: INVITE
Example: set apply-to-methods INVITE+REFER
apply-to-responses: Specifies whether to apply header value changes to SIP requests or requests and responses. Set to no to apply changes only to requests. Set to yes to apply to responses as well. If yes, you must set the response code to which it applies. Create additional altered-body profiles to change multiple response types.
Default: no Values: no | yes responseCode
Example: set apply-to-responses yes 200
session-persistent: Specifies to which messages in a session SessionManager should apply changes made with this object. When enabled, the ME applies any TO, FROM, or REQUEST URI changes to the first and all subsequent messages in a session. When disabled, the default, the system applies the changes only to the first message in the session.
Default: disabled
Values: enabled | disabled
Example: set session-persistent enabled
cseq: Secondary property. Sets a mechanism to further filter which SIP messages have the header expression modifications applied. If cseq is set to zero (the default), the ME applies the changes to all SIP messages. If set to any other value, the system only applies the changes to SIP messages having a CSEQ field that matches that value.
Default: 0
Example: set cseq 100
apply-to-dialog: Allows you to configure where to apply these options for a session. The following are valid values:
Default: both
Values: inbound: Apply to inbound dialog only.
outbound: Apply to outbound dialog only.
both: Apply to both inbound and outbound dialogs.
Example: set apply-to-dialog inbound
Sets the authentication mode to use on this SIP session. For authentication services that involve remote servers (such as RADIUS and DIAMETER), you must configure these servers on the ME using the either the radius-group or diameter-group configuration objects. For the directory authentication mechanism, you must first configure the directory service in the directory configuration object.
config vsp default-session-config authentication config vsp policies session-policies policy name rule name session-config authentication config vsp dial-plan dial-prefix entryName session-config authentication config vsp dial-plan route name session-config authentication config vsp dial-plan source-route name session-config authentication config vsp session-config-pool entry name authentication
mode: Sets the type of authentication the system uses for this SIP session. Optionally, you can set whether the authentication applies to inbound-only (enabled) or inbound and outbound (disabled) traffic.
Default: none disabled
Values: none <enabled | disabled>: The system performs no authentication.
Local <enabled | disabled>: The system uses the username and password configured in the user object for authentication.
RADIUS <enabled | disabled> <radiusGroupReference>: The system performs RADIUS authentication according to the configuration specified in the radius-group object.
DIAMETER <enabled | disabled><diameterGroupReference>: The system performs DIAMETER authentication according to the configuration specified in the diameter-group object.
Directory <enabled | disabled><directoryReference>: The system expects the user credentials that are specified in the directory service that you supply.
Example: set mode radius enabled ”vsp radius-group boston1”
session-starter-only: Specifies which requests the system challenges. When disabled (the default), if authentication is enabled the system challenges all requests in a session. When enabled, the system only challenges the first request in a session.
Default: disabled
Values: enabled | disabled
Example: set session-starter-only enabled
handle-challenge-locally: Sets whether a challenge is handled locally. When enabled, the system terminates the original challenge response (either 401 Unauthorized or 407 Proxy Authentication Required) and generates a new request with the authentication information. When disabled, the system forwards the 401/407 response back to the UAC.
Default: disabled
Values: enabled | disabled
Example: set handle-challenge-locally enabled
challenge-response-code: Sets the response code that the system sends when it terminates the original challenge response (either 401 Unauthorized or 407 Proxy Authentication Required). This code is only applied when the handle-challenge-locally property is enabled.
Default: 401
Values: 401 | 407
Example: set challenge-response-code 407
apply-to-methods: Specifies which message types to authenticate. This setting is used by the registration-throttling property of the route and source-route registration plan objects to define which message types require authentication.
When you modify this value, the system overwrites the current setting with only the message types you specify. For example, if set to the default and you enter INVITE, the system only authenticates INVITE messages. Enter multiple types separated by a plus sign (+) with no spaces.
Default: INVITE+REGISTER+BYE
Example: set apply-to-methods INVITE+REFER+REGISTER
exclude-scheme-in-called: Secondary property. Specifies which portions of the TO URI that the ME uses for authentication. If set to false, the system authenticates the full TO URI. If true, the system uses only the User and Host portions of the TO URI.
Default: false
Values: true | false
Example: set exclude-scheme-in-called true
initial-challenge-stale: Specifies whether the stale parameter is included in authentication challenges, per RFCs 2069 and 3261.
Default: true
Values: true: Includes stale=”true” in the challenge.
false: Includes stale=”false” in the challenge.
none: The stale parameter is not included in the challenge.
Example: set initial-challenge-stale none
Sets the type of authorization the system performs for matching sessions. Using this object, you set the protocol the system uses to get authorization data: either none, theroute-server engine, or WSDL. This data results in a list of routing options for the call. For route-server, you must configure the route-server master service and, for intercluster lookups, a diameter client and server. For WSDL, you configure an external-services > policy-group.
config vsp default-session-config authorization config vsp policies session-policies policy name rule name session-config authorization config vsp dial-plan dial-prefix entryName session-config authorization config vsp dial-plan route name session-config authorization config vsp dial-plan source-route name session-config authorization config vsp session-config-pool entry name authorization
mode: Sets the method to use for authorization data retrieval.
Default: none
Values: None: The system performs no route-server lookup. This is the equivalent of administratively disabling the route-server service for matching calls.
Local: The system performs intracluster route-server lookup; it sends the route lookup request to the system hosting the route-server master service.
WSDL <policyGrpReference><true | false>: The system sends the request to the external services policy server specified in the policy-group configuration. Optionally, you can specify whether to send SIP message headers and/or content (both default to false) with the request.
Diameter <diameterGrpReference>: The system sends the route request to the server specified in the diameter-group object configuration. This is typically only used for intercluster lookup configurations.
RADIUS: The The ME sends a request to the RADIUS server with the to-URL and from-URL in the request. The RADIUS server responds with information that the ME uses to create session-configs that are applied to the session.
Example: set mode Diameter ”vsp diameter-group rsserver”
always-perform-lookup: Specifies whether the system should do an authorization lookup (if configured to do so with the mode property). If set to true, the default, the system retrieves authorization data regardless of other configuration settings. If set to false, the system first uses internal logic to determine whether session handling data can be derived from other sources (e.g., location cache or dial plan). Set this to false, for example, when handling two locally registered phones calling each other.
Default: true
Values: true | false
Example: set always-perform-lookup false
apply-to-methods: Specifies to which message types the system applies authorization processing.
When you modify this value, the system overwrites the current setting with only the message types you specify. For example, if set to the default and you enter INVITE, the system only authorizes INVITE messages. Enter multiple types separated by a plus sign (+) with no spaces.
Default: INVITE
Example: set apply-to-methods INVITE+REGISTER
sequence: Select an existing sequence to use for querying the route server. This is a sequence you must have configured in the vsp > route-server-config > route-server-sequence object.
Default: There is no default setting
Example: set sequence query1
Sets the body types that are allowed and/or prohibited during the session. This functionality is initiated (if configured) when the ME receives a SIP message that contains more than one type in the body portion of the message (when the Content Type header indicates that the message has multiple and mixed parts.) This object defines which types survive and which are deleted from the message.
Select a body type: application or text: and then a specific subtype. Use the custom MIME type to either add a body part other than application or text, or to use a subtype for application or text that is not preconfigured. To allow only a single body type, set allowed-body-part to the desired type and set blocked-body-type to any.
config vsp default-session-config bodypart-type config vsp policies session-policies policy name rule name session-config bodypart-type config vsp dial-plan dial-prefix entryName session-config bodypart-type config vsp dial-plan route name session-config bodypart-type config vsp dial-plan source-route name session-config bodypart-type config vsp session-config-pool entry name bodypart-type
allowed-body-part: Sets the body part types allowed during the session. Re-execute the command for each type you want to allow.
Default: The default subtype setting for audio, video, and application is any; there is no default setting for custom-mime-type. Values: audio subType | video subType | application subType | custom-mime-type mimeType subType
Example: set allowed-body-part custom-mime-type application safe-trade
blocked-body-part: Sets the body part types to prohibit during the session. Any body sections that contain this type are removed from the message before forwarding. Re-execute the command for each type you want to block.
Default: The default subtype setting for audio, video, and application is any; there is no default setting for custom-mime-type. Values: audio subType | video subType | application subType | custom-mime-type mimeType subType
Example: set blocked-body-part video mpls
move-bp-headers: Specifies how to handle headers when there are changes to the message body. If enabled, when a message that has multiple parts is reduced into a single body part, the system moves the remaining body part header into the message header.
Default: disabled
Values: enabled | disabled
Example: set move-bp-headers enabled
Provides third-party conferencing, allowing a third-party participant, such as an emergency service endpoint, to be added to a call in progress. The third-party endpoint may or may not be registered with the ME. Note that this feature differs from the monitor-group three-way calling feature. When using monitor groups, calls can only be listened to. Through this object a third-party can join a conversation.
config vsp default-session-config media call-monitoring config vsp policies session-policies policy name rule name session-config media call-monitoring config vsp dial-plan dial-prefix entryName session-config media call-monitoring config vsp dial-plan route name session-config media call-monitoring config vsp dial-plan source-route name session-config media call-monitoring config vsp session-config-pool entry name media call-monitoring
admin: Specifies whether the system initiates conference calling. When enabled, and there is a session configuration match, the system makes a call to the specified third party. Upon answering, the third party is conferenced into the active call. You must also enable the media anchor property to use this feature.
Default: enabled
Values: enabled | disabled
Example: set admin disabled
monitor-uri: Specifies the third party endpoint that is to be conferenced in to the active call. Note that only one call monitoring endpoint is supported per session configuration.
Default: There is no default setting
Example: set monitor-uri http://localEmerg.Services.net/monitor0
Specifies how matching calls are mapped to calling groups and the group reference from which they pick up their provisioning. See Configuring Calling Group Objects for complete information on calling groups.
config vsp default-session-config calling-group-settings config vsp policies session-policies policy name rule name session-config calling-group-settings config vsp dial-plan dial-prefix entryName session-config calling-group-settings config vsp dial-plan route name session-config calling-group-settings config vsp dial-plan source-route name session-config calling-group-settings config vsp session-config-pool entry name calling-group-settings
type: Sets the calling-group association.
Default: There is no default setting
Values: dynamic: Creates a calling group, and names it based on the AOR of the device. The new group then inherits the settings of the group referenced in the calling-group property.
configured: Assigns the matching AOR to the group referenced with the calling-group property. This assignment overrides any other calling group assignment (e.g., registration-plan > route match).
Example: set type dynamic
calling-group: Specifies the referenced group for calling group parameters. If type is set to configured, it is the group that the AOR joins. If set to dynamic, it is the group from which the new groups settings are inherited.
Default: There is no default setting
Example: set calling-group ”vsp calling-groups groupEast”
Specifies the list of either additional codecs to allow and/or verify beyond the ME internal list, or allows overriding the parameters of the ME internal codecs. The ME codec name list is pre-populated with internal codecs by their official Session Description Protocol (SDP) tag. Enter a predefined SDP tag to open the codec object. (Type a question mark at the command line to see the list of predefined tags.) Or, to add a new codec, specify its SDP tag.
payload-type: Specifies how the ME calculates the RTP payload.
Default: automatic
Values: automatic: The system determines the payload based on the SDP rtpmap attribute, indicating the payload-type of the specified codec tag. For nearly all cases, this should be left as automatic
manual<hex-value>: Use this if the codec is always known to be constant or if the SIP client does not correctly use the rtpmap attribute. Specify the hexadecimal value that specifies the media payload type.
none: N/A.
Example: set payload-type none
packet-size: Specifies how the ME calculates RTP packet size.
Default: automatic; if set to manual, the default packet size is 100 bytes and cannot be changed
Values: automatic: The system determines the packet size based on the codec tag. For nearly all cases, this should be left as automatic as codecs use RFCs and other SDP parameters (number of channels, bit rate, sampling frequency, etc.) to determine the correct minimum and maximum sizes.
manual<min-size><max-size>: Use this if a client does not specify a parameter correctly, or this is not an internal codec. Specify the minimum and maximum sizes in bytes.
none: Disables packet-size checking.
Example: set packet-size manual 100 100
packet-rate: Specifies how the ME calculates the maximum RTP packet rate..
Default: automatic; if set to manual, the default size is 100 packets per second
Values: automatic: The system determines the packet rate based on the codec tag. For nearly all cases, this should be left as automatic as codecs use RFCs and other SDP parameters (number of channels, bit rate, sampling frequency, etc.) to determine the correct maximum rate.
manual <max-size>: Use this if a client does not specify a parameter correctly, or this is not an internal codec. Specify the maximum rate in packets per second.
none: Disables packet-rate checking.
Example: set packet-rate manual 50
Specifies the CODEC and the parameter to apply to it in the a=fmtp line of the SDP.
vsp session-config-pool entry codec-specific-parameters codec-parameters vsp default-session-config entry codec-specific-parameters codec-parameters
param: Specifies the maximum number of frames per second (FPP) for audio CODECs that the ME advertises. Enter the type of codec followed by the max-fpp.
Default: There is no default codec. The default max-fpp is 24.
Values: any
g722
g7221
g723
g729
g729a
gsm
pcma
pcmu
string
Values: Min: 0 / Max: 255
Example: set param audio g729 35
The codec-payload-type-bindings configures a binding between a codec name and a payload type. Without any codec-payload-type-bindings configured, the ME uses a default DTMF payload type of 101.
This configuration element is set when you want to change the default DTMF payload type offered by the ME. This property takes precedence over the default of 101. Codec-payload-type-bindings is used when the ME generates its own SDP for outgoing calls. The ME generates its own SDP for features like file-play or when the ME is in a Delayed-Offer/Early-Offer network.
config vsp default-session-config in-media-normalization codec-payload-type-bindings config vsp default-session-config out-media-normalization codec-payload-type-bindings config vsp session-config-pool entry <name> in-media-normalization codec-payload-type-bindings config vsp session-config-pool entry <name> out-media-normalization codec-payload-type-bindings
Adds an a=fmtp line to SDP. Certain endpoints require CODEC parameters to be specified; this object allows you to add the necessary information. Note that if SDP arrives with the a=fmtp line specified for a CODEC set with this object, the original line remains. Use this object in conjunction with the sdp-regeneration object to override the line. First, use sdp-regeneration to remove the received a=ftmp line from SDP, then use this object to add the line back in with the desired parameters.
config vsp default-session-config codec-specific-parameters config vsp policies session-policies policy name rule name session-config codec-specific-parameters config vsp dial-plan dial-prefix entryName session-config codec-specific-parameters config vsp dial-plan route name session-config codec-specific-parameters config vsp dial-plan source-route name session-config codec-specific-parameters config vsp session-config-pool entry name codec-specific-parameters
Specifies where the ME derives the content of the CONTACT header in 3xx (response/redirect) messages that it receives. The CONTACT headers in a 3xx response tell the recipient of the request (which caused the redirect to be generated) where the request can be sent that will yield a final response. This ”alternate location” information is sent back to the UAC via the Contact: header(s) in the 3xx response. Set this object in cases when you want the ME to forward the 3xx message with a CONTACT header containing something other than the ME itself. (Note, however, that in that case, the UA does not return the updated information to the ME device.)
For example, if the user property is set to to-uri, the ME replaces the user field of the CONTACT header with data from the user field of the outgoing TO header in the 3xx response.
config vsp default-session-config contact-uri-settings-3xx-response config vsp policies session-policies policy name rule name session-config contact-uri-settings-3xx-response config vsp dial-plan dial-prefix entryName session-config contact-uri-settings-3xx-response config vsp dial-plan route name session-config contact-uri-settings-3xx-response config vsp dial-plan source-route name session-config contact-uri-settings-3xx-response config vsp session-config-pool entry name contact-uri-settings-3xx-response
user: Specifies how to derive the value of the User field (the resource located at host) of the CONTACT header.
Default: contact-uri
Values: request-uri: Uses the value from the incoming REQUEST URI
to-uri: Uses the value from the incoming TO URI
from-uri: Uses the value from the incoming FROM URI
next-hop: Uses the IP address of the next-hop server
omit: Leaves the field blank
string: Writes the specified string to the field
Example: set user from-uri
user-prefix: Appends the specified string to the beginning of the User field of the CONTACT header. Use this, for example, if you need to append a ”1” to a phone number for an outside call.
Default: There is no default setting
Example: set user-prefix 1
host: Specifies how to derive the value of the Host field (the host providing SIP resource) of the CONTACT header.
Default: next-hop-address
Values: cxc-address: Uses the ME IP address as the host
public-address: Uses the public address for a UAC behind a firewall or the UAC address if it is not behind a firewall
original-address: The host field is not modified
next-hop-address: Uses the IP address of the next-hop server. However, this value is typically used only for the
string: Writes the specified string to the field
Example: set host original-address
port: Specifies how to derive the value of the Port field (where the request is to be sent) of the CONTACT header.
Default: omit
Values: cxc-local-port: Uses the port number that the system transported the call over
original-port: The port field is not modified
omit: Leaves the field blank
string: Writes the specified string to the Port field of the 3xx CONTACT header
Example: set port original-port
transport: Specifies the derivation of the transport type for the Transport field of the CONTACT header.
Default: omit
Values: next-hop-transport: Uses the method used by the next-hop server
original-transport: The transport field is not modified
omit: Leaves the field blank
UDP | TCP | TLS: Sets the transport field to the selected protocol
Example: set transport original-transport
add-maddr: When enabled, the ME adds a maddr URI parameter.
Default: disabled
Values: enabled | disabled
Example: set add-maddr enabled
Specifies where the ME derives the content of the CONTACT header from when it forwards a message to a UAC. For example, if the user property is set to to-uri, the ME replaces the User field of the CONTACT header with data from the User field of the incoming TO header. The inbound leg of the session is the portion from the ME to the call initiator (UAC).
Note that this modification does not apply to REGISTER requests. To make changes to the headers of a REGISTER request, use the properties of the registration-plan object.
config vsp default-session-config contact-uri-settings-in-leg config vsp policies session-policies policy name rule name session-config contact-uri-settings-in-leg config vsp dial-plan dial-prefix entryName session-config contact-uri-settings-in-leg config vsp dial-plan route name session-config contact-uri-settings-in-leg config vsp dial-plan source-route name session-config contact-uri-settings-in-leg config vsp session-config-pool entry name contact-uri-settings-in-leg
user: Specifies how to derive the value of the User field (the resource located at host) of the CONTACT header.
Default: contact-uri
Values: request-uri: Uses the value from the incoming REQUEST URI
to-uri: Uses the value from the incoming TO URI
from-uri: Uses the value from the incoming FROM URI
next-hop: Uses the IP address of the next-hop server
omit: Leaves the field blank
string: Writes the specified string to the field
Example: set user from-uri
host: Specifies how to derive the value of the Host field (the host providing SIP resource) of the CONTACT header.
Default: cxc-address
Values: cxc-address: Uses the ME IP address as the host
public-address: Uses the public address for a UAC behind a firewall or the UAC address if it is not behind a firewall
original-address: The host field is not modified
next-hop-address: Uses the IP address of the next-hop server. However, this value is typically used only with the contact-uri-settings-3xx-response object.
string: Writes the specified string to the field.
Example: set host original-address
port: Specifies how to derive the value of the Port field (where the request is to be sent) of the CONTACT header.
Default: cxc-local-port
Values: cxc-local-port: Uses the port number that the system transported the call over
original-port: The port field is not modified
omit: Leaves the field blank
string: Writes the specified string to the field
Example: set port original-port
transport: Specifies the derivation of the transport type for the Transport field of the CONTACT header.
Default: next-hop-transport
Values: next-hop-transport: Uses the method used by the next-hop server
original-transport: The transport field is not modified
omit: Leaves the field blank
UDP | TCP | TLS: Sets the transport field to the selected protocol
Example: set transport original-transport
add-maddr: Specifies whether to include the MAddr parameter in the CONTACT header. If enabled, the system adds its own IP address as the MAddr parameter. If disabled, the system replaces the HOST with its IP address.
Default: enabled
Values: enabled | disabled
Example: set add-maddr disabled
use-incoming-contact: Determines the basis for creating the CONTACT header in an outbound message. When enabled, the system first copies the content of the inbound header to build the outbound header. Using the inbound header content, the system then applies any further changes defined in this contact-uri-settings-in-leg object.
Default: disabled
Values: enabled | disabled
Example: set use-incoming-contact enabled
from-user-contact-uri: This property is not applicable to the inbound leg of a connection.
Default: disabled
Values: enabled | disabled
Example: set from-user-contact-uri enabled
registration-plan-precedence: Sets whether or not the system applies the CONTACT header modifications specified within this object. When set to true, the system does not apply changes to messages if a registration-plan is present. When set to false, all message types are changed.
Default: true
Values: true | false
Example: set registration-plan-precedence false
add-other-params: Specifies whether the system maintains additional parameters in incoming CONTACT headers. When enabled, the system allows any additional parameters that were received in the CONTACT header to remain in the new CONTACT header when it rewrites it as a result of matching this session config. Additional (or ”other”) parameters are those found after the URI. For example, in the header ”Contact: <sip:johnD@10.1.1.1:5060 udp>;team,” ”team” is the other parameter and remains in the new CONTACT header. When disabled, the system removes additional parameters.
Default: disabled
Values: enabled | disabled
Example: set add-other-params enabled
always-include-contact: Sets the system verify whether there is a contact header present in each message. When enabled, the system checks to ensure that there is a Contact header present. If there is not, it creates one. The content of the header is derived from the other properties in this object. When disabled, the system does not check for the presence of the Contact header.
Default: disabled
Values: enabled | disabled
Example: set always-include-contact enabled
Specifies where the ME derives the content of the CONTACT header from when it forwards a message to a UAS. For example, if the user property is set to to-uri, the ME replaces the User field of the CONTACT header with data from the User field of the outgoing TO header. The outbound leg of the session is the portion from the ME to the call responder (UAS).
Note that this modification does not apply to REGISTER requests. To make changes to the headers of a REGISTER request, use the properties of the registration-plan object. Also, if you have enabled enum-apply-request-result-to-contact in the dial-plan > normalization object, you must set this object so that the Contact header is not overwritten on the outbound side. To do so, set use-incoming-contact to enabled, set host, port, and transport to use their original values, and set user to the Contact URI.
config vsp default-session-config contact-uri-settings-out-leg config vsp policies session-policies policy name rule name session-config contact-uri-settings-out-leg config vsp dial-plan dial-prefix entryName session-config contact-uri-settings-out-leg config vsp dial-plan route name session-config contact-uri-settings-out-leg config vsp dial-plan source-route name session-config contact-uri-settings-out-leg config vsp session-config-pool entry name contact-uri-settings-out-leg
user: Specifies how to derive the value of the User field (the resource located at host) of the CONTACT header.
Default: contact-uri
Values: request-uri: Uses the value from the incoming Request URI
to-uri: Uses the value from the incoming To URI
from-uri: Uses the value from the incoming From URI
contact-uri: Uses the value from the incoming Contact URI. Select this value to preserve changes made to the Contact header through the dial-plan > normalization object.
omit: Leaves the field blank
string: Writes the specified string to the field
Example: set user from-uri
host: Specifies how to derive the value of the Host field (the host providing SIP resource) of the CONTACT header.
Default: cxc-address
Values: cxc-address: Uses the ME IP address as the host
public-address: Uses the public address for a UAC behind a firewall or the UAC address if it is not behind a firewall
original-address: The host field is not modified. Select this value to preserve changes made to the Contact header through the dial-plan > normalization object.
next-hop-address: The IP address of the next-hop server. However, this value is typically used only for the contact-uri-settings-3xx-response object.
string: Writes the specified string to the field
Example: set host original-address
port: Specifies how to derive the value of the Port field (where the request is to be sent) of the CONTACT header.
Default: cxc-local-port
Values: cxc-local-port: Uses the port number that the system transported the call over
original-port: The port field is not modified. Select this value to preserve changes made to the Contact header through the dial-plan > normalization object.
omit: Leaves the field blank
string: Writes the specified string to the field
Example: set port omit
transport: Specifies the derivation of the transport type for the Transport field of the CONTACT header.
Default: next-hop-transport
Values: next-hop-transport: Uses the method used by the next-hop server
original-transport: The transport field is not modified. Select this value to preserve changes made to the Contact header through the dial-plan > normalization object.
omit: Leaves the field blank
UDP | TCP | TLS: Wets the transport field to the selected protocol
Example: set transport original-transport
add-maddr: Specifies whether to include the MAddr parameter in the CONTACT header in an outbound message. If enabled, the system adds its own IP address as the MAddr parameter. If disabled, the system replaces the HOST with its IP address.
Default: enabled
Values: enabled | disabled
Example: set add-maddr disabled
use-incoming-contacts: Determines the basis for creating the CONTACT header in an outbound message. When enabled, the system first copies the content of the inbound header to build the outbound header. Using the inbound header content, the system then applies any further changes defined in this contact-uri-settings-out-leg object. Set this property to enabled to preserve changes made to the Contact header through the dial-plan > normalization object.
Default: disabled
Values: enabled | disabled
Example: set use-incoming-contacts enabled
from-user-contact-uri: Specifies whether the system uses the location cache to derive the CONTACT header when forwarding a message. When disabled, the default, the CONTACT URI is derived from the FROM header of the original message. When this property is enabled, the system does a location cache lookup on the received FROM URI. If the system finds an entry, it uses the server-side contact (found in the entry) as the CONTACT URI for the outbound message.
Default: disabled
Values: enabled | disabled
Example: set from-user-contact-uri enabled
registration-plan-precedence: Sets whether or not the system applies the CONTACT header modifications specified within this object to REGISTER requests. When set to true, the system does not apply changes to REGISTER messages. When set to false, all message types are changed.
Default: true
Values: true | false
Example: set registration-plan-precedence false
add-other-params: Specifies whether the system maintains additional parameters in incoming CONTACT headers. When enabled, the system allows any additional parameters that were received in the CONTACT header to remain in the new CONTACT header when it rewrites it as a result of matching this session config. Additional (or ”other”) parameters are those found after the URI. For example, in the header ”Contact: <sip:johnD@10.1.1.1:5060 udp>;team,” ”team” is the other parameter and remains in the new CONTACT header. When disabled, the system removes additional parameters.
Default: disabled
Values: enabled | disabled
Example: set add-other-params enabled
always-include-contact-header: Sets the system verify whether there is a contact header present in each message. When enabled, the system checks to ensure that there is a Contact header present. If there is not, it creates one. The content of the header is derived from the other properties in this object. When disabled, the system does not check for the presence of the Contact header.
Default: disabled
Values: enabled | disabled
Example: set always-include-contact-header enabled
Provides CSTA-to-OCI, -OCS, or - communications for enterprises using a BroadWorks, Cisco, or Avaya call manager and Microsoft OCS. The ME supports third-party call control (3PCC) for any phones connected to the 3PCC server. See Configuring Third-Party Call Control server objects for a complete description of 3PCC and information on configuring 3PCC servers.
Note that to make this application work in addition to the ME configuration, you must also point the CSTA SIP traffic at the ME so that it acts as a CSTA gateway. See the Oracle Communications Session Services Configuration Guide for more information.
Every device is mapped to a unique terminal ID. When a device logs into MOC, the ME records the terminal ID. It is not uncommon for phone address to be mapped to multiple devices, and therefore, associated with multiple IDs. Because the ME always selects the active device when initiating a session, there must be some logic configured to identify the active device so that only that device is used when an outbound call is made through MOC. This is useful, for example, when a home and work phone are mapped to the same URI. The terminal-select-dial property (in this object) configures the ME to call a specified number (and play a recorded file). When a user picks up in response to the call, the device used to answer is noted as the active terminal and calls to that URI are forwarded to that active device. Note that after an ID has been established through this configuration, if a different device answers a call, the ME uses the new device for the current call and then resumes use of the established device for future calls. (To change the active terminal setting, use the jtapi-control action.)
The ME allows you to use the multiple partition feature of Cisco CallManager. A single phone with a single phone number (DN) can be mapped to two or more partitions, where each partition can map to a individual line on the phone. The primary partition will indicate the line to use for outbound calls, typically this will be line 1. See the Cisco online documentation, Partitions and Calling Search Spaces, for complete information on Cisco partitions.
config vsp default-session-config csta-settings config vsp policies session-policies policy name rule name session-config csta-settings config vsp dial-plan dial-prefix entryName session-config csta-settings config vsp dial-plan route name session-config csta-settings config vsp dial-plan source-route name session-config csta-settings config vsp session-config-pool entry name csta-settings
mode: Specifies the 3PCC server type that the ME is connecting to the Microsoft OCS application. By selecting a server type, the ME acts as a translation device, converting CSTA traffic from that server type to a format the 3PCC server can recognize. Enter the type and a reference to the configured 3PCC server.
Default: none
Values: none <serverReference>: The ME Engine does not provide 3PCC services.
internal <serverReference>: The system acts as the PBX, resulting in phones registering with the ME Session Controller Media Engine device. This mode only works with phones registered directly to the ME device.
broadworks <serverReference>: The ME converts CSTA traffic to either OCI or OCS traffic, depending on the type property setting of the referenced BroadWorks server.
cisco <serverReference>: The ME converts CSTA traffic to for processing by the referenced Cisco server.
avaya <serverReference>: The ME converts CSTA traffic to for processing by the referenced Avaya server.
loopback <serverReference>: The ME creates a loopback session to the OCS for testing.
Example: set mode broadworks ”vsp enterprise 3pcc-servers broadworks-csta-server BWocs”
terminal-select-dial: See Identifying the Active Divide for information on the use of this property. In the example below, the system looks changes calls in the form of tel:+1508xxxyyyy to 1508xxxyyyy@callme.com. The result is the number the system dials to play the file. The entry in the From URI field is displayed as the caller ID.
Default: any Values: any | disabled | once-at-login <toURIexp><toURIreplace><fromURIfile> | action-driven <toURIexp><toURIreplace><fromURIfile>
Example: set terminal-select-dial once-at-login ^tel:(\+)?((1?508) [0-9]{7}).*$\2callme.com gday.wav
lcs-transport: Sets the transport protocol used to communicate with the third-party server. For a secure connection and to support CSTA failover operations, set transport to TLS and include a reference to a certificate on the system.
Default: any Values: any | UDP | TCP | TLS <certificateReference>
Example: set lcs-transport tls ”vsp tls certificate nnos-e.abc.com
default-partition: Sets which partition this session configuration applies to (controls). This feature only applies to Cisco CallManager partitions; see Partitions and Calling Search Spaces in the Cisco online documentation for more information on partitions.
Note that you can also set the default partition using the set-default-partition option of the jtapi-control action. The action setting overrides the values set with this property.
For more information regarding configuring regular expressions and replacement strings, see Using Regular Expressions.
Default: automatic
Values: automatic: Use the first address from the list retrieved from the switch.
specified <value>: Use the partition that you specify.
derived <regEx><replacement>: Use the partition name found in the Active Directory number, and derived by a regular expression rule. The AD number is in the form tel:+number:partition.
Example: set default-partition derived ”(?ms).*<RequestSystemStatus.*tel:\
+[0-9]*;(.*)PT””\1PT”
Adds a custom data field to the callCreated, callConnected, and callTerminated events. This object defines the content of the field.
config vsp session-config-pool entry third-party-call-control custom-event-fields
config vsp default-session-config third-party-call-control custom-event-fields
named-variable-entry: Specifies the content of the custom data field added to the accounting record. This is in the form variable=value.
Default: There is no default setting
Example: set named-variable-entry variable=\s
custom-events-grouping-string: Secondary property. The characters used to associate an event's variable and values.
Default: =
Example: set custom-events-grouping-string :
custom-event-delimiter: Secondary property. The characters used to separate group custom event entries.
Default: ;
Example: set custom-event-delimiter \
Configures the DNS client process and how the client communicates with the DNS service (resolver). See DNS Service Resolver and Server Objects for information on the ME DNS service.
config vsp default-session-config dns-client-settings config vsp policies session-policies policy name rule name session-config dns-client-settings config vsp dial-plan dial-prefix entryName session-config dns-client-settings config vsp dial-plan route name session-config dns-client-settings config vsp dial-plan source-route name session-config dns-client-settings config vsp session-config-pool entry name dns-client-settings
admin: Specifies whether this DNS client configuration entry is applied to calls matching the session configuration.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
client-timeout: Specifies how long in milliseconds the client process waits for the DNS service (resolver) to respond. If this timer expires, the service continues to look for the entry, and on finding it, writes it to its own and to the client cache. When the client next queries for that same address, the response will come from the cache.
Default: 2000
Example: set client-timeout 3000
server-names: Specifies which server(s) the DNS service should use to resolve requests that originate from calls matching this session configuration. Enter a reference to a server name that you configured with the resolver server object. If you do not configure any server-names with this property, all configured servers are used.
Default: There is no default setting
Example: set server-names ”vsp dns resolver server name dns1”
routing-last-resort-dns: Specifies whether the system should do a DNS lookup when it cannot determine where to forward a call based on the dial plan, registration plan, location cache, or policy. If enabled, the system does a DNS lookup on servers configured with the DNS resolver server object. If disabled, the system does not do a DNS lookup and returns, by default, a ”404 not found” message to the caller. You can change the response code and string using the sip-settings dns-fail-response-code and -string properties.
Default: disabled
Values: enabled | disabled
Example: set routing-last-resort-dns enabled
add-nnos-domain: Secondary property. Specifies whether to add the configured domain to a single-label query. If disabled, no domain is added and the setting in the use-cxc-domain-in-search and additional-search-domains properties of the resolver object are applied. If enabled, the ME appends the name set with the domain-name property of the static-stack-settings object to a single-label query, making it a FQDN. The other resolver settings do not apply.
Default: disabled
Values: enabled | disabled
Example: set add-nnos-domain enabled
routing-lookup-type: Secondary property. Sets the method of server location. This property is applicable when the routing-last-resort-dns property is enabled.
Default: NAPTR+SRV+A
Values: NAPTR+SRV+A: A NAPTR lookup, an SRV lookup on the information returned, and an A lookup from those results. It also does a lookup on the original domain name.
NAPTR+SRV: A NAPTR lookup, an SRV lookup on the information returned, and an A lookup from those results. There is no lookup on the original domain name.
SRV+A: An SRV lookup, followed by an A lookup on names returned by SRV. Finally, it does a lookup on the original domain name.
SRV: An SRV lookup, followed by an A lookup on names returned by SRV.
A: A lookup on the original domain name.
Example: set routing-lookup-type SRV
ras-settings: Sets the configuration for scenarios when the ME is communicating with an external H.323 gatekeeper. (This property is only applicable if the server-type property is set to h323-gatekeeper.) When the ME registers on behalf of a client, these settings allow the systems to exchange registration, admission, and status (RAS) messages.
Default: The default settings are indicated in each field description
Values: registration TTL: Sets the frequency, in seconds, of the system putting forward reregistrations. The default is 3600 seconds
registration retries: Sets the number of attempts the system makes to register a client. before abandoning the request. A value of zero allows unlimited retries. The default setting is 5 retries.
endpoint alias: Assigns a string to identify the system to the gatekeeper. This string should be configured on the gatekeeper as well so that it can recognize calls from the ME.
supported prefix: Sets the value for gatekeepers that need digits prepended to a number. This requirement would be set in the gatekeeper dial plan. Usually, this field is left blank.
reregister on UNREGISTER: Sets whether the system tries to reregister a client after having received an UNREGISTER from the gatekeeper. If this is enabled, the ME tries to reregister the client up to the number f times specified with the registration-retires field. The default setting is disabled.
gatekeeper call routing: Sets whether the system provides support for gatekeeper-routed calls. When true, the ME does provide that support. The default setting is false.
Example: set ras-settings 5200 8 nnos-e-1 gk1 1! enabled true
Sets whether matching calls should be handled without limitation. When this object is administratively enabled, matching calls are not subject to emission and admission controls.
config vsp default-session-config emergency-settings config vsp policies session-policies policy name rule name session-config emergency-settings config vsp dial-plan dial-prefix entryName session-config emergency-settings config vsp dial-plan route name session-config emergency-settings config vsp dial-plan source-route name session-config emergency-settings config vsp session-config-pool entry name emergency-settings
Sets the 3GPP policy configuration to apply to SIP sessions.
config vsp default-session-config endpoint-management
config vsp session-config-pool entry name endpoint-management
admin: Enables or disables endpoint management.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
park-incoming-calls: Enable or disable automatically parking incoming calls on the ME.
Default: disabled
Values: enabled | disabled
Example: set park-incoming-calls enabled
park-call-greeting: Specify the audio file to be played when an incoming call is parked.
Default: There is no default setting
Example: set park-call-greeting greet1.wav
terminate-after-greeting: Enables or disables the ME terminating a call after parking it and playing a greeting.
Default: enabled
Values: enabled | disabled
Example: set termiante-after-greeting disabled
Configures call control events as well as named-variables and user-specified channels.
config vsp session-config-pool entry name event-settings
config vsp default-session-config event-settings
call-control-events: Enables or disables call control events.
Default: enabled
Values: enabled | disabled
Example: set call-control-events disabled
media-control-events: Enables or disables media control events.
Default: enabled
Values: enabled | disabled
Example: set media-control-events disabled
channel: Specifies channels to be used for events generated for this session. Named variables can be added into channels by using the % character to delimit the name of the variable to expand.
Default: call
Example: set channel %$event.requestID%
allow-event-group-events: Secondary property. Enables or disables the ME to send events via the legacy event-group object.
Default: enabled
Values: enabled | disabled
Example: set allow-event-group-events disabled
inbound-sip-messages: Secondary property. Configures events for incoming SIP messages.
outbound-sip-messages: Secondary property. Configures events for outgoing SIP messages.
MESSAGE-events: Secondary property. Enables or disables the MESSAGEevents group. This event group exists for backwards compatibility only. The SipMessageEvent class should be used instead.
Default: disabled
Values: enabled | disabled
Example: set MESSAGE-events enabled
named-variable-entries: Inserts named-variables into events. Before selecting named-variables to enter into events, you must configure them in either the session-config > named-variables object or via the named-variables-add action.
Default: There is no default setting
Example: set named-variable-entries my-variable my-variable-name
allow-empty-value: Secondary property. When enabled, the ME allows you to create named-value-pair entries in events without any corresponding values. The ME then looks in the configured named variable table to find a value to include in the event. If the value is empty and this property is enabled, the ME writes out the variable in the event but leaves it blank.
Default: enabled
Values: enabled | disabled
Example: set allow-empty-value disabled
event-filter: Configures a filter which allows you to specify the event classes that this session is allowed to emit.
Default: There is no default setting
Example: set event-filter call-create
Enables recording of file transfers on the ME.
config vsp default-session-config file-transfer config vsp policies session-policies policy name rule name session-config file-transfer config vsp dial-plan dial-prefix entryName session-config file-transfer config vsp dial-plan route name session-config file-transfer config vsp dial-plan source-route name session-config file-transfer config vsp session-config-pool entry name file-transfer
anchor: Enables or disables anchoring, which defines whether the system is used as an intermediary for traffic. When enabled, all file transfers pass through the ME device. If disabled, transfers circumvent the ME device. (The system would still have a record of the transfer, however, because it keeps records of all SIP transactions.) You must enable anchoring to use the recording or virus scanning features.
Default: disabled
Values: enabled | disabled
Example: set anchor enabled
record: Enables or disables recording of file transfers on the system in this SIP call session. The file is stored on the system and forwarded to the SIP call recipient. Anchoring must be enabled to use the recording feature.
Default: disabled
Values: enabled | disabled
Example: set record enabled
max-filesize: Configures the maximum file size, in bytes, allowed in a transfer through the ME device. If the size limit is exceeded, the file is dropped.
Default: 1073741824
Values: Min: 1 / Max: 1073741824
Example: set max-filesize 50000000
allow-non-default-ports: Specifies whether or not the system is limited in choice of ports when anchoring a file transfer. When enabled, the default, the system can use any port. Leave this setting for interoperability with LCS 2005. When disabled, the system can only use the default port that is specified by Microsoft.
Default: enabled
Values: enabled | disabled
Example: set allow-non-default-ports disabled
t120-anchor: Enables anchoring of the application sharing and whiteboard features found in the Office Communicator 2005 client (based on ITU T.120). If you enable this feature, you must also enable the anchor property of the media object. If you are anchoring federated traffic, you must also set the sip-settings > lcs-compatibility bit to compensate for a bug in the OC 2005 client that does not accept Content-Type headers with the subtype ”SDP” specified in uppercase. Set the bit to 0x010032e3.
Default: disabled
Values: enabled | disabled
Example: set t120-anchor enabled
Configures the ring pattern for SIP calls from the ME to the endpoint. The ME selects ring destinations based on information in the address of record (AOR) from the locations services database.
You can also configure an ordered set of rules (metrics) to influence the final server (or next hop) selection for outbound calls with the outbound-arbiter-rule property. This may be necessary if you have configured multiple possible next-hop devices using the peer object. The arbitration calculation rules apply to all outbound devices to determine the destination. This set of rules takes precedence over any arbitration decision made through the dial-plan arbiter configuration. If a dial-plan references a session-config in which this object outbound-arbiter-rule is configured, the ME ignores the dial-plan arbiter configuration and uses this one instead.
config vsp default-session-config forking-settings config vsp policies session-policies policy name rule name session-config forking-settings config vsp dial-plan dial-prefix entryName session-config forking-settings config vsp dial-plan route name session-config forking-settings config vsp dial-plan source-route name session-config forking-settings config vsp session-config-pool entry name forking-settings
forking-type: Sets the ME forwarding behavior when it receives a call INVITE. This is the method in which the system forwards the call to the endpoint. In order for this property to work properly, the outbound-arbiter-rule parameter must be configured.
Default: none
Values: none: The system forwards the call to the latest binding it finds for the Request URI in the location service.
sequential: The call is forwarded to each binding for each AOR stored for a destination, one at a time, until it is successful. If the system receives a fail message, busy, or timeout, it tries the next AOR. The delay between trying each device is set with the session-provisional-timeout property of the sip-settings object.
parallel: The call is forwarded to each binding for all AORs for the destination.
redirect: The redirect call statistics are used as inputs in the algorithm. This must be the configured forking-type in order for the 302 redirect feature to work.
Example: set forking-type parallel
max-hunt: Specifies the number of destinations to try when the system is configured to do sequential forking. Setting this value prevents the creation of a forking loop in the event that a server redirects a call to another server in the listed destinations.
Default: 50
Example: set max-hunt 50
outbound-arbiter-rule: Enters rules into the arbiter configuration. Enter as many rules as you wish. If you do not set any rules, the system uses the settings of the dial-plan arbiter (or factory defaults if the dial-plan also has no arbiter configuration). If you select least-cost, you can optionally set a maximum (or unlimited) value for call cost. It you select trunk-qos, you can optionally select a previously configured class-of-service. This property is required for the forking-settings properties to work properly.
See the Routing algorithm options table for a description of the selections.
Default: There is no default setting
Example: set outbound-arbiter-rule least-cost 15
max-arbitration-options: Secondary property. Specifies the number of potential destinations to consider when applying a rule. The smaller of this and max-hunt takes effect when the final destinations are determined.
Default: unlimited
Values: Min: 0 / Max: 7294967295
Example: set max-arbitration-options 50000
Specifies what derives the content of the fields of the FROM URI when the ME transmits a message. For example, if the user property is set to from-uri, the ME replaces the user field of the FROM URI with data from the user field of the incoming FROM URI. If set to omit, the user field is left blank. Or, you can enter any string that you want placed in the user field. See Altering URIs for information on replacement options.
config vsp default-session-config from-uri-specification config vsp policies session-policies policy name rule name session-config from-uri-specification config vsp dial-plan dial-prefix entryName session-config from-uri-specification config vsp dial-plan route name session-config from-uri-specification config vsp dial-plan source-route name session-config from-uri-specification config vsp session-config-pool entry name from-uri-specification
user: Specifies how to derive the value of the user field of the FROM URI.
Default: from-uri Values: request-uri | to-uri | from-uri | omit | next-hop | local | omit-phone-context | string
Example: set user request-uri
host: Specifies how to derive the value of the host field of the FROM URI.
Default: from-uri Values: request-uri | to-uri | from-uri | omit | next-hop | next-hop-domain | local-ip | string
Example: set host omit
port: Specifies how to derive the value of the port field of the FROM URI.
Default: from-uri Values: request-uri | to-uri | from-uri | omit | string
Example: set port omit
display: Specifies how to derive the value of the display field of the FROM URI.
Default: from-uri Values: request-uri | to-uri | from-uri | omit | next-hop | string
Example: set display next-hop
user-agent-aware-display-translation: Specifies whether or not to apply the displayname-character-set-info mappings to matching calls. When enabled, the system checks the display name for accented characters. If found, it does a location cache lookup on the destination. If that is successful, it checks the user agent found in the location cache against the configured mappings and performs any necessary translations.
Default: disabled
Values: enabled | disabled
Example: set user-agent-aware-display-translation enabled
transport: Specifies the value of the transport field of the FROM URI. In addition to using the value from other fields of the incoming URI, you can set the transport method to UDP, TCP, or TLS.
You cannot enter a string for this property.
Default: from-uri Values: request-uri | to-uri | from-uri | omit | UDP | TCP | TLS | string
Example: set transport omit
use-param: Specifies whether the User parameter in the FROM URI of the SIP header is maintained or removed when the system forwards a message. If set to keep, the message is forwarded with the parameter as it was received. If set to omit, the entire user=param is removed from the TO URI.
Default: omit
Values: omit | keep
Example: set use-param keep
user-truncate-non-digits: Specifies whether to remove non-digits from the User portion of the FROM URI in INVITE messages. When enabled, the system removes all non-digits.
Default: disabled
Values: enabled | disabled
Example: set user-truncate-non-digits enabled
uri-parameter <name><value>: Appends the specified user parameter and value to the FROM URI for matching calls. For example, the example below would result in a FROM URI that looked similar to:
<sip:spot@fun.com;BTG=trunk1>
You can control how and when the new parameter is added using the append options. Select append-always to have the parameter added to all matching calls. Select append-if-does-not-exist to add the name and value only if it does not already exist in the URI, preventing the possibility of duplicate parameters. Select overwrite-existing to replace any existing parameter in the FROM URI with the configured name and value, updating instead of appending to the parameter. You can append multiple user parameters.
Default: There is no default setting
Example: set uri-parameter BTG trunk1
header-parameter: Adds a parameter string to the SIP header (outside of the SIP URI). Use this string, for example, to identify the source of a call or group destinations. You can add as many header parameters as required. Use the format name=value. To add multiple parameters use the format name=value;name=value...
Default: There is no default setting
Example: set header-parameter OLI=70
add-oli-tag: Specifies the number of digits to copy from the User portion of the From header to create an Originating Line Information (OLI) tag. The OLI tag provides information on the class of service available for the call. A value of 0 disables adding the tag.
Default: 0
Example: set add-oli-tag 2
copy-charge-uri-user: Specifies whether to copy the User portion of the Charge URI to the User portion of the From URI. When enabled, the system changes the content of the User portion, for example, for billing purposes.
Default: disabled
Values: enabled | disabled
Example: set copy-charge-uri-user enabled
strip-digits: Specifies the number of digits to strip from the User portion of the From URI. Use this, for example, if the original INVITE contains extra digits that would be problematic to the downstream server. Digits are removed beginning at the string that immediately follows the sip: or sips: portion.
Default: 0
Example: set strip-digits 2
prepend-digits: Specifies a string to prepend to the User portion of the From URI. The string can be comprised of up to 128 ASCII characters.
Default: There is no default setting
Example: set prepend-digits 011
Creates a tag (group name) that can be stored in the location cache during registration. When the ME receives a REGISTER and applies a matching session configuration, it saves out the zero or more groups (configured with this object) that are associated with that session config. When the ME receives an INVITE destined for a registered phone, a location cache lookup results in return of all the stored group names, which can then be used to further refine the selection of the applicable session config for the INVITE. You can use this feature, for example, to control outbound settings that are specific to a type of phone, such as the encryption type or CODEC preferences.
config vsp default-session-config group-settings config vsp policies session-policies policy name rule name session-config group-settings config vsp dial-plan dial-prefix entryName session-config group-settings config vsp dial-plan route name session-config group-settings config vsp dial-plan source-route name session-config group-settings config vsp session-config-pool entry name group-settings
The H.225 settings configuration object allows you to configure H.225 on the ME.
config vsp session-config-pool entry h225-settings config vsp default-session-config h224-settings
false-start: If enabled, the ME accepts inbound H.323 fast start calls and includes fast start in SETUP messages for outbound H.323 calls. The calls fall back to slow if fast start is unsuccessful.
Default: enabled
Values: enabled | disabled
Example: set false-start disabled
manual-ringback: If enabled, the ME prohibits remote ringback. When this property is disabled, SIP to H.323 calls attempt to open an audio channel for remote ringback.
Default: enabled
Values: enabled | disabled
Example: set manual-ringback disabled
use-inbound-call-settings: When enabled for an H.323 to H.323 call, the ME uses inbound H.323 call settings for H.323 outbound calls.
Default: disabled
Values: enabled | disabled
Example: set use-inbound-call-settings enabled
fwd-progress-as-alerting: When enabled, the ME sends an Alerting message instead of a Progress message.
Default: disabled
Values: enabled | disabled
Example: set fwd-progress-as-alerting enabled
default-terminal-type: Identifies the ME terminal type for MSD.
Default: 60
Values: Min: 0 / Max: 4294967295
Example: set default-terminal-type 75
multiple-calls: When enabled, the ME allows calls to share an H.225 connection.
Default: disabled
Values: enabled | disabled
Example: set multiple-calls enabled
maintain-connection: When enabled, the ME keeps an H.225 connection open after calls are cleared.
Default: disabled
Values: enabled | disabled
Example: set maintain-connection enabled
conn-idle-timeout: Specifies the maximum lifetime of an idle H.225 connection. A value of 0 indicates an idle connection should never timeout.
Default: 3600
Values: Min: 300 / Max: 65535
Example: set conn-idle-timeout 2500
h323-user-alias: Specifies the source and destination address type in Setup, Alerting, Connect, ARQ, and LRQ messages.
Default: none
none
dialedDigits
h323ID
urlID
emailID
Example: set h323-user-alias urlID
call-alerting-timeout: The maximum number in seconds the ME waits for Alerting message after sending a SETUP. The call clears if this timeout is reached.
Default: 4
Values: Min: 0 / Max: 4294967295
Example: set call-alerting-timeout 500
call-establishment-timeout: The maximum number in seconds the ME waits for an H.323 call to be established. The call clears if this timeout is reached.
Default: 60
Values: Min: 0 / Max: 4294967295
Example: set call-establishment-timeout 75
end-session-timeout: The maximum number of seconds the ME waits after sending a ReleaseComplete before call resources are reclaimed.
Default: 15
Values: Min: 0 / Max: 4294967295
Example: set end-session-timeout 30
h245-establish-timeout: The maximum number, in seconds, the ME waits for an H245 connection to be established. The call clears if this timeout is reached.
Default: 1
Values: Min: 0 / Max: 4294967295
Example: set h245-establish-timeout 5
reinvite-type: Secondary property. Indicates if the ME should use Terminal Capability Set or Extended Fast Connect messages to reconfigure media channels.
Default: emptyTermCapSet
Values: emptyTermCapSet | extendedFastConnect
Example: set reinvite-type extendedFastConnect
use-progress-inband: When enabled, inband ring information from the inbound H.323 call-leg is propagated to the outbound call-leg.
Default: enabled
Values: enabled | disabled
Example: set use-progress-inband disabled
fwd-retrieve-no-tx: When true, the ME does not pause remotetransmitted if media information is 0.0.0.0.
Default: true
Values: true | false
Example: set fwd-retrieve-no-tx false
use-server-connection: Secondary property. specifies whether the ME creates a new, or uses an existing, TCP connection. If true, the ME uses a TCP connection created by the remote gateway instead of creating a new outbound TCP connection. Use this property for a remote H.323 gateway using connection sharing for its H.225 traffic. (It uses a single TCP connection for multiple calls.)
Default: true
Values: true | false
Example: set use-server-connection false
enum-lookup-called-party: When enabled, the ME performs an ENUM lookup of the called number before making an outbound H.323 call.
Default: disabled
Values: enabled | disabled
Example: set enum-lookup-called-party enabled
enum-domain: The domain used for ENUM lookups.
Default: 164.arpa
Example: set enum-domain 12025551234
enum-returnednaptr-replace: Secondary property. Enter a regexp and a replacement value. When configured, if the ME performs an ENUM dip for an inbound H.323 call, the regexp and replacement string are applied to the result of the ENUM lookup. That then becomes the called party identifier.
Default: There is no default setting
Example: set enum-returnednaptr-replace (*)\?
session-duration-max: Sets the maximum duration of an H.323 call, in seconds. A value of 0 (the default) indicates there is no maximum lifetime.
Default: 0
Values: Min: 0 / Max: 1000000
Example: set session-duration-max 1000
The H.245 settings configuration object allows you to configure H.245 on the ME.
config vsp session-config-pool entry h245-settings config vsp default-esssion-config h245-settings
h245-tunnel: When enabled, the ME attempts to use an H.225.0 connection for H.245 traffic. The use of H.245 tunneling depends on indication from both H.323 terminals and gateways.
Default: enabled
Values: enabled | disabled
Example: set h245-tunnel disabled
early-h245: The ME does not support early H.245.
Default: notunnel
Values: notunnel: The ME ignores the early H.245 and completes the call setup using slowstart.
reject: The ME rejects the call.
Example: set early-h245 reject
wait-for-remote-tcs: When true, the ME waits to receive a Terminal Capability Set message before advertising its capabilities. When false, the ME issues a TCS message after a slowstart call is connected.
Default: true
Values: true | false
Example: set wait-for-remote-tcs false
clc-when-pausing-remote: Secondary property. When true, the ME closes its TX channels when pausing the remote H.323 terminal.
Default: false
Values: true | false
Example: set clc-when-pausing-remote true
send-msd-when-unpausing-remote: Secondary property. Specifies whether the ME conducts MSD when using TCS to unpause a remote H.323 gateway.
Default: false
Values: true | false
Example: set send-msd-when-unpausing-remote true
use-h450-hold-retrieve: When enabled, the ME uses H.450 supplemental service PDUs for holds and retrieves.
Default: enabled
Values: enabled | disabled
Example: set use-h450-hold-retrieve disabled
sip-h323-dtmf-translate <sip-dtmf-type><h323-dtmf-type>: Sets preferences for H.323-SIP DTMF interworking for a particular H.323 trunk.
Default: inband
Example: set sip-h323-dtmf-translate RFC2833 H245SIGNAL
codec-selection: Secondary property. Indicates how the ME chooses converged codecs.
Default: remote
Values: -none: No codec is being used.
-local: Use the highest preference common codec seen in SIP SDP.
-remote: Use the highest preference common codec in remote TCS.
-followMSD: Use the result of MDT to decide.
Example: set codec-selection local
map-ptime-to-fpp: Secondary property. When set to true, the ME uses SDP ptime parameter to set max-frames-per-packet codec value in Terminal Capability Set. Ptime and FPP are not equivalent, however, this allows compatibility in some IW scenarios.
Default: false
Values: true | false
Example: set map-ptime-to-fpp true
map-fpp-to-ptime: Secondary property. When true, the ME uses max-frames-per-packet codec value in Terminal Capability Set to set SDP ptime parameter. Ptime and FPP are not equivalent, however, this allows compatibility in some interworking scenarios.
Default: false
Values: true | false
Example: set map-fpp-to-ptime true
add-equivalent-codecs: Secondary property. When true, the ME adds equivalent codecs to Terminal Capability Set. The currently supported case is G729 present in SDP which would add both G729 and G729A in TCS.
Default: false
Values: true | false
Example: set add equivalent-codecs true
Specifies how to generate a SIP From header from an H.323 SETUP message. When The ME receives a message from an H.323 server via the server that contains this configuration object, it creates the From header using the parameters of this object. The From header is made up of four components defined here--scheme:user@host.suffix.
config vsp session-config-pool entry h323-to-sip-fromheader-spec config vsp default-session-config h323-to-sip-fromheader-spec
scheme: Specifies the Scheme to use in the From (or To) header.
Default: sip
Values: sip: Use the SIP scheme
tel: Use the tel scheme
omit: Leaves the field blank. Select this if the upstream server uses the correct scheme in the H.323 SETUP message and you do not want that value changed.
string: Enters the specified string in the scheme field
Example: set scheme tel
user: Specifies the origin of the User field content to use in the From (or To) header.
Default: calling-number
Values: calling-number: The value in the originating phone number
h323-id: The value from the incoming h323ID alias
url-id: The value from the incoming url ID
email-id: The value from the incoming email ID
omit: Leaves the field blank
string: Enters the specified string in the User field
Example: set user h323-id
host: Specifies the origin of the Host field content to use in the From (or To) header.
Default: h323gw-domain
Values: h323-id: The value from the incoming h323ID alias
url-id: The value from the incoming url ID
email-id: The value from the incoming email ID
h323gw-domain: Omits the value configured for the H323 gateway
omit: Leaves the field blank
string: Enters the specified string in the User field
Example: set host omit
suffix: Specifies the suffix to add to the From (or To) header. Enter a suffix or select omit to let the system derive the field from the SETUP message.
Default: omit Values: omit | string
Example: set suffix omit
use-anon: When enabled, as the H.323 process builds the SIP From: header for a received H.323 SETUP message it will do the following:
Use anonymous as the user portion of the URI if after applying h323-to-sip-fromheader-spec config the user portion is empty
Use the IP address of the H.323 gateway which transmitted the SETUP as the host portion of the URI if, after applying h323-to-sip-fromheader-spec config, the host portion is empty
This guarantees a valid From: header URI will exist when sent to the SIP process. When set to false there is some chance an incomplete URI could be passed to SIP.
Default: false
Values: true | false
Example: set use-anon true
Specifies how to generate a SIP To header from an H.323 SETUP message. When the ME receives a message from an H.323 server via the server that contains this configuration object, it creates the To header using the parameters of this object. The To header is made up of four components defined here: scheme:user@host.suffix.
The ME supports Type of Service (ToS) marking for H.323 packets. The ToS value determines the quality of service that a call receives. The ToS byte in the IP header is used to mark packets for special consideration during routing. When the ME forwards a packet marked with a ToS value, this information is used by downstream routers to prioritize packet forwarding or perform other quality of service mechanisms.
When the ME receives an in-leg H.323 session with a ToS value set, you have the option to either forward this value to the out-leg session or override the in-leg ToS.
Additionally, when the ME receives an in-leg SIP session, you can either preserve the initial ToS value or override it in the out-leg during IW translation to H.323.
You configure in-leg and out-leg ToS settings for H.323 packets under this object.
For messages sent via TCP, the ECN field (the least significant two bits of the ToS) can neither be preserved nor overwritten by the ME. For these sessions, the ECN field in outgoing packets is always marked as zero, regardless of the incoming or overwritten value.
Additionally, if TLS over TCP is supported via H.235, the ToS value cannot be preserved, but it can be overwritten to the same ECN limitation as TCP.
Sets whether events are sent to a third-party server via .
config vsp default-session-config handle-publish config vsp policies session-policies policy name rule name session-config handle-publish config vsp dial-plan dial-prefix entryName session-config handle-publish config vsp dial-plan route name session-config handle-publish config vsp dial-plan source-route name session-config handle-publish config vsp session-config-pool entry name handle-publish
Assigns an action to a matching response code. If the ME receives a response to a SIP request (typically a UAS) that is within the 400-999 range, you can specify how the ME responds (forwarding the response message or re-sending the INVITE to a different peer or route). The purpose of this feature is to determine the destination to which the ME forks a call.
The ME operates in the following manner. If:
The handle-response property is configured for the session-config and there is a matching response code, the ME takes the configured action. If the response code does not match, the ME uses the default action (try-next-peer).
The handle-response property is not configured for the session-config, the ME uses the handle-response setting in the server configuration (if it exists).
When the ME receives a the specified response code from a call, it takes one of the following actions:
try-next-peer: The ME forwards the message to the next server within a route.
try-next-route: The ME forwards the message to the route that is the next most-specific.
forward: The ME returns the response to the originator of the message.
A call can match more than one route, and each route may have more than one destination. For example, a call may route to destinations A and B. Destination A may have routes A1, A2, and A3. Destination B may have routes B1 and B2. If the current destination is A1, and a handle-response code match occurs with a setting of try-next-peer, the ME forwards the message to A2. If the setting is try-next-route, the ME forwards the message to B1.
config vsp default-session-config handle-response config vsp policies session-policies policy name rule name session-config handle-response config vsp dial-plan dial-prefix entryName session-config handle-response config vsp dial-plan route name session-config handle-response config vsp dial-plan source-route name session-config handle-response config vsp session-config-pool entry name handle-response
entry: Specifies the action the system should take when it receives a specific response code from a call matching this session-config. Enter a code, and set a handling pattern.
Default: There is no default setting
Values: try-next-peer: The system forwards the message to the next server for a route
try-next-route: The system forwards the message to the route that is the next most-specific
forward: The system returns the response to the originator of the message
Example: set entry handle-response 404 try-next-route
Modifies the User portion of the specified header. This object uses the same methodology as the dial-plan > normalization object.
config vsp default-session-config header-settings header-normalization number config vsp policies session-policies policy name rule name session-config header-settings header-normalization number config vsp dial-plan dial-prefix entryName session-config header-settings header-normalization number config vsp dial-plan route name session-config header-settings header-normalization number config vsp dial-plan source-route name session-config header-settings header-normalization number config vsp session-config-pool entry name header-settings header-normalization number
admin: Enables or disables this configuration entry.
Default: enabled
Values: enabled | disabled
Example: set admin disabled
destination: Specifies the header to be normalized. The system makes changes to the User field of the destination URI. Changes are applied to all messages types identified in the apply-to-methods property.
Default: There is no default setting
Example: set destination Diversion
value: Sets the type of normalization that the system applies to outgoing calls to a provider (to the USER field of the destination URI). See User Normalization Properties for property setting options and descriptions.
Default: none (no normalization is applied)
Example: set value replace-prefix 866
apply-to-methods: Specifies the message type to which the system applies header value changes. The system then changes the specified URI according to the settings of the value property of this object.
When you modify this value, the system overwrites the current setting with only the message types you specify. For example, if set to the default and you enter NOTIFY, the system only modifies NOTIFY messages. Enter multiple types separated by a plus sign (+) with no spaces.
Default: INVITE
Example: set apply-to-methods INVITE+CONTACT
apply-to-response: Specifies whether to apply header value changes to SIP requests or requests and responses. Set to no to apply changes only to requests. Set to yes to apply to responses as well. If yes, you must set the response code to which it applies. Create additional altered-body profiles to change multiple response types.
Default: no Values: no | yes responseCode
Example: set apply-to-response yes 200
session-persistent: Specifies to which messages in a session the ME should apply changes made with this object. When enabled, the ME applies any TO, FROM, or REQUEST URI changes to the first and all subsequent messages in a session. When disabled, the default, the system applies the changes only to the first message in the session.
Default: disabled
Values: enabled | disabled
Example: set session-persistent enabled
cseq: Secondary property. Sets a mechanism to further filter which SIP messages have the header expression modifications applied. If cseq is set to zero (the default), the ME applies the changes to all SIP messages. If set to any other value, the system only applies the changes to SIP messages having a CSEQ field that matches that value.
Default: 0
Example: set cseq 100
Configures the ME to remove, or to remove and replace the content of fields from the SIP header. In addition, you can identify header types to specifically allow or block. The allowed-headers and blocked-headers properties apply the following rules:
The From, To, CSeq, and Call-ID headers are required and cannot be blocked.
The allowed list overrides settings of the blocked list. If a header is explicitly allowed, it cannot then be blocked using the blocked list.
If a header name does not match a value in either list, it is allowed.
The ME accepts regular expressions for an entry; all special characters apply.
config vsp default-session-config header-settings config vsp policies session-policies policy name rule name session-config header-settings config vsp dial-plan dial-prefix entryName session-config header-settings config vsp dial-plan route name session-config header-settings config vsp dial-plan source-route name session-config header-settings config vsp session-config-pool entry name header-settings
allowed-header: Sets the SIP headers that should be explicitly allowed to remain in the SIP message. You can enter any number of header names by re-executing the command. See the Purpose for applicable rules.
Default: There is no default setting
Example: set allowed-header Via
blocked-header: Sets the SIP headers that should be explicitly removed from the SIP message. You can enter any number of header names by re-executing the command. See the Purpose for applicable rules.
Default: There is no default setting
Example: set blocked-header .*
apply-allow-block-to: Sets whether the allow and block properties of this object apply to requests only or requests and responses. When disabled, changes apply only to requests. When enabled, the default, changes apply to requests and responses.
Default: requests-and-responses
Values: requests | responses | requests-and-responses
Example: set apply-allow-block-to responses
pAssert-mode: Secondary property. Sets whether to strip the number in the P-Asserted-Identity field from the SIP header. When enabled, the system replaces the value in the From field with the value from the P-Asserted-Identity field for the outbound call leg. (Note that the system maintains the original From field value in the Contact field.)
Default: disabled
Values: enabled | disabled
Example: set pAssert-mode enabled
header-to-strip: Secondary property. Configures the system to strip the value of the specified field. Enter a SIP header field name.
Default: There is no default setting
Example: set header-to-strip Remote-Party-ID
apply-to-allow-block-to-dialog: Specifies whether the allow and block properties of this object apply to a specific dialog or not.
Default: both
Values: inbound: Apply to the inbound dialog only
outbound: Apply to the outbound dialog only
both: Apply to both inbound and outbound dialogs
Example: set apply-to-allow-block-to-dialog inbound
sip-manipulation: Specify the configured sip-manipulation you want to associate with this header-setting. Configure the sip-manipulation in the sip-manipulation-pool > sip-manipulation object.
Default: There is no default setting
Example: set sip-manipulation sipmanip1
Sets a preference for CODECs, influencing the ME ordering of them in the SDC on the inbound leg of a call. The ME removes those CODECs with a zero priority from the SDP. CODEC preferences do not cause the ME to add a CODEC to the SDP, but to remove and/or reorder existing CODECs according to their priority. The ME places a CODEC whose priority is not specified in its original order, just ahead of the known auxiliary CODECs (e.g., telephone-events). This is not a mechanism to add CODECs into an SDP, only to order those that are already there (via transcoding or from the original offer/answer).
config vsp default-session-config in-codec-preferences config vsp policies session-policies policy name rule name session-config in-codec-preferences config vsp dial-plan dial-prefix entryName session-config in-codec-preferences config vsp dial-plan route name session-config in-codec-preferences config vsp dial-plan source-route name session-config in-codec-preferences config vsp session-config-pool entry name in-codec-preferences
preferences<media-type>[codec][priority]: Assigns a priority to a given CODEC for inbound audio or video sessions.
Default: There are no default settings
Values: media type: Select for audio or video. The associated CODEC (subtype) is preferred according to the priority for that media type.
codec: The CODEC to which the priority applies. Use the question mark character at the command line to see a list of available CODECs, or enter any CODEC name.
priority: Sets a preference for the CODEC. The lower the number, the more preferred the CODEC. Assigning a priority value of zero disables the CODEC for the session. The system removes these CODECs before sending the SDP offer or answer. (0-100)
Example: set preferences video g729 1
Configures the ME's in-leg DTMF method preferences.
config vsp default-session-config in-dtmf-preferences config vsp session-config-pool entry in-dtmf-preferences
admin: Specifies whether or not this DTMF preference list is applied to calls matching this session configuration.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
preferences: Allows you to configure supported dtmf-types and assign them with a priority to determine the ME's preferences.
Default: audio 1
First select a DTMF method. The available DTMF methods are:
Values: audio
rfc-2833
sip-info-dtmf
sip-info-dtmf-relay
sip-notify
h245-alphanumeric
h245-signal
q931
Then assign it a priority. This can be from 0-100. A value of 0 means the method is not supported. The lower the priority, the more preferred the DTMF method.
Example: set preferences rfc-2833 2
The ME can be configured to translate one DTMF method to another. Through this object, you can control the length of play and pause time and volume for the digits that the ME plays on the in-leg.
config vsp default-session-config in-dtmf-settings config vsp session-config-pool entry out-dtmf-settings
digit-volume: Specifies the volume setting for the DTMF tones. The digit volume is measured in decibel (dB) of the measured power referenced to one milliwatt, measured at a zero transmission level point. The smaller the dBm0, the louder the volume.
Default: -20
Values: Min: -36 / Max: 0
Example: set digit-volume -15
digit-duration: Specifies the length of time, in milliseconds, that the ME plays each DTMF digit.
Default: 750
Values: Min: 100 / Max: 10000
Example: set digit-duration 1000
min-digit-duration: Specifies the minimum length of time, in milliseconds, that the ME plays each DTMF digit. If a DTMF event has a duration less than this value, the digit-duration property overrides the duration and is used to play the DTMF event.
Default: 60
Values: Min: 5 / Max: 100
Example: set min-digit-duration 75
max-digit-duration: Specifies the maximum length of time, in milliseconds, that the ME plays each DTMF digit. If a DTMF event has a duration greater than this value, the digit-duration property overrides the duration and is used to play the DTMF event.
Default: 2000
Values: Min: 100 / Max: 10000
Example: set max-digit-duration 3000
inter-digit-duration: Specifies the length of time, in milliseconds, that the ME pauses between playing each digit.
Default: 250
Values: Min: 0 / Max: 1000
Example: set inter-digit-duration 500
pause-duration: Specifies the length of time, in milliseconds that the ME pauses when it encounters a comma character in the conference code. The comma is a special character in the conference code that indicates a specified time the ME must wait before playing the next tone.
Default: 3000
Values: Min: 500 / Max: 10000
Example: set pause-duration 4000
minimum-duration: Specifies the minimum time, in milliseconds, between detecting RFC-2833 events.
Default: 60
Values: Min: 0 / Max: 1000
Example: set minimum-duration 100
as-audio: Specifies whether the ME sends audio or DTMF packets to the conference server when representing conference code tones. When true, the ME encodes the sound in the current CODEC. When false, the ME sends DTMF packets.
Default: true
Values: true | false
Example: set as-audio false
Secondary object. Controls the method used for forwarding DTMF tones in a call. The two supported methods are via the signaling stream using SIP INFO messages or via a DTMF packet that is in compliance with RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. Using this object, you can configure the ME to:
Forward packets in the form they arrived.
Pick them out of the RTP stream and send them in a SIP INFO message (if they arrived as DTMF packets).
Extract them from a SIP INFO message and send them as DTMF packets (if the SIP INFO message contained a DTMF body).
Inbound DTMF translation applies to the segment from the initiator to the ME device.
config vsp default-session-config in-dtmf-translation config vsp policies session-policies policy name rule name session-config in-dtmf-translation config vsp dial-plan dial-prefix entryName session-config in-dtmf-translation config vsp dial-plan route name session-config in-dtmf-translation config vsp dial-plan source-route name session-config in-dtmf-translation config vsp session-config-pool entry name in-dtmf-translation
info: Specifies the method to use for forwarding DTMF tones that were received in a SIP INFO message. If set to info, the system forwards the message as it was received (in an INFO message). If set to rfc-2833, the system extracts the DTMF body from the INFO message and sends the content via DTMF packets.
Default: info
Values: info | rfc-2833
Example: set info rfc-2833
drop-info: Specifies whether to drop the SIP INFO message if the info property is set to rfc-2833. If set to true, the system drops the INFO packet and only sends the DTMF packets. If set to false, the system sends both.
Default: false
Values: true | false
Example: set drop-info true
rfc-2833: Specifies the method to use for forwarding DTMF tones that were received in DTMF packets. If set to rfc-2833, the system forwards the message as it was received (in DTMF packets). If set to info, the system extracts the DTMF packets from the RTP stream and sends the content via a SIP INFO message. The system sends one INFO message per event detected.
Default: rfc-2833
Values: info | rfc-2833
Example: set rfc-2833 info
drop-rfc-2833: Specifies whether to drop the DTMF packets if the rfc-2833 property is set to info. If set to true, the system drops the RFC 2833 packets and only sends the SIP INFO message. If set to false, the system sends both.
Default: false
Values: true | false
Example: set drop-rfc-2833 true
info-dtmf-body: Specifies the body type to use in a SIP INFO message when converting from RFC 2833 format. When using dtmf, the message body contains just single character (the digit that was pressed). When set to dtmf-relay, the body contains the single character plus duration data.
Default: dtmf-relay
Values: dtmf-relay | dtmf
Example: set info-dtmf-body dtmf
timeout-rfc-2833: Secondary property. Sets the number of milliseconds the system waits before sending a SIP INFO message if it does not detect the end of the event. The timer is started at the start of an event. This property only applies when the forwarding method has been changed from rfc-2833 to info, and is used in the event that when monitoring DTMF, the system does not detect an event end.
Default: 1000
Example: set timeout-rfc-2833 1500
Configures inbound server-side acousting echo cancellation for devices running telephony applications that do not have functional on-board echo cancellation.
config vsp session-config-pool entry name in-echo-cancellation-settings
config vsp default-session-config in-echo-cancellation-settings
admin: Enables or disables echo cancellation on inbound call-legs.
Default: disabled
Values: enabled | disabled
Example: set
platform-delay: Although the maximum echo tail allowed by the echo cancellation algorithm is 200 ms, it is possible that the actual echo tail for certain devices is longer. This parameter allows you to account for any extra delay in addition to the 200 ms. For example, if your total acoustic delay is 500 ms, set echo-tail to 200 ms and platform-delay to 300 ms. This covers the entire 500 ms delay.
Default: 0
Values: Min: 0 / Max: 65535
Example: set platform-delay 100
echo-tail: The amount of time, in milliseconds, that it takes for the person speaking to hear his or her own echo. This is also known as echo delay. The maximum echo tail supported by the ME's algorithm is 200 ms.
Default: 16
Values: Min: 0 / Max: 200
Example: set echo-tail 150
noise-reduction: Allows you to reduce background noise. A value of 0 means there is no noise reduction and 5 is the maximum reduction.
Default: 2
Values: Min: 0 (no noise reduction) / Max: 5
Example: set noise-reduction 4
Sets the parameters for inbound encrypted media sessions when the ME is anchoring a call. This is the portion from the initiator to the ME device. With this object you set the encryption requirements on the call, and the encryption method used.
config vsp default-session-config in-encryption config vsp policies session-policies policy name rule name session-config in-encryption config vsp dial-plan dial-prefix entryName session-config in-encryption config vsp dial-plan route name session-config in-encryption config vsp dial-plan source-route name session-config in-encryption config vsp session-config-pool entry name in-encryption
mode: Specifies the encryption requirements on the incoming call (from endpoint to the ME device). The method of encryption used is determined by the type property.
Default: none
Values: none: The system disables the encryption put forth by the incoming endpoint (i.e, it responds ”no” to the encryption portion of the authentication handshake.) If the outbound endpoint requires encryption, the call is dropped.
pass-thru: The system passes cryptographic parameters through the box and does not participate in RTP encryption. This method renders some advanced media services unusable (in particular, recording, announcements, transcoding, call monitoring, RTP stats, media verification, and RTCP generation). When using this option, set mode to pass-thru in both in-encryption and out-encryption.
allow: If the incoming endpoint offers encryption to the ME device, the system answers with it. If the endpoint does not offer encryption, the system does not answer with or initiate encryption.
follow: If the outbound endpoint offers encryption, the system offers encryption (the type set by policy) to the inbound endpoint.
offer: The system offers encryption to the inbound endpoint when the session is first established.
reoffer: The system offers encryption to the inbound endpoint whether the message is an INVITE or a REINVITE. This setting is most applicable when an endpoint issues a REINVITE, and encryption was not required with the original INVITE. In this case, the system will again offer encryption when forwarding the message.
require: The call must come in with encryption specified or the system drops it.
Example: set mode allow
type: Sets the type of encryption on inbound sessions. In choosing a type, the system uses the encryption expected by that device or application.
Default: RFC-3711
Values: Linksys: Uses SIP INFO messages to exchange mini-certificates and exchange a symmetric key. The media encryption is similar to RFC-3711, but done with AES-128 Countermode and with HMAC MD5 authentication. See the linksys action for more certificate information.
RFC-1889: Use encryption as defined in RFC 1889, RTP: A Transport Protocol for Real-Time Applications. This mode is used for compatibility with Windows Messenger and Microsoft Office Communicator, neither of which currently support RFC-3711 encryption. Instead, it uses a DES-CBC encryption of the entire UDP payload (including RTP headers) with no authentication.
RFC-3711: Use encryption as defined in RFC 3711, The Secure Real-time Transport Protocol (SRTP).
Example: set type Linksys
require-tls: Specifies the requirements of the signaling protocol for the call inbound leg. It defines whether the system offers SRTP over a non-secure (TCP or UDP) signaling connection. The action of this property depends on the setting of the mode property.
In most cases, this property does not need to be modified because the system does not consider the in-leg transport (only whether or not crypto was offered). However, set this property to true to ensure that the system does not offer crypto to a client on the in-leg that is not using TLS.
Default: false
Values: true: The system only offers encryption when talking to a TLS client. If TLS and SRTP are required (mode is set to require), the system fails calls going to TCP/UDP clients. If the mode property is set to offer or follow, the system forwards the call without SRTP.
false: The system offers SDP messages according to the mode setting without regard for the signaling transport. This allows keys to be exchanged in an insecure message.
Example: set require-tls true
priority-AES-128-CM-HMAC-SHA1-32: Secondary property. Sets a preference for 32-bit SHA1 authentication tags on incoming calls. The system supports (offers) both 32- and 80-bit authentication tags on ingress. A value of 0 disables support for the 32-bit tag.
Default: 1
Values: Min: 1 (most preferred) / Max: 5
Example: set priority-AES-128-CM-HMAC-SHA1-32 2
priority-AES-128-CM-HMAC-SHA1-80: Secondary property. Sets a preference for 80-bit SHA1 authentication tags on incoming calls. The system supports (offers) both 32- and 80-bit authentication tags on ingress. A value of 0 disables support for the 80-bit tag. To enable, set a value between 1 (most preferred) and 5.
Default: 2
Values: Min: 1 (most preferred) / Max: 5
Example: set priority-AES-128-CM-HMAC-SHA1-80 1
mki-length: Secondary property. Provides support for the optional Master Key Identifier bit defined in RFC 3711, The Secure Real-time Transport Protocol (SRTP). The value specify sets the number of bytes in the MKI. The system then sends the negotiated identifier of that length indicating which master key to use for decryption with each SRTP packet. Note that the endpoint must support this option.
Default: 0
Values: Min: 0 / Max: 4
Example: set mki-length 2
mikey-offer-location: Secondary property. Controls where in the SDP the system stores the MIKEY offer (the ”a=key-mgmt:mikey” line) when it is made. If MIKEY is offered to the system, it puts the MIKEY answer in the location where the offer was located. Set the location as the media descriptor or session level.
Default: session
Values: session | media-descriptor
Example: set mikey-offer-location media-descriptor
mikey-time-tolerance: Secondary property. Controls where in the SDP the system stores the MIKEY offer (the ”a=key-mgmt:mikey” line) when it is made. If MIKEY is offered to the system, it puts the MIKEY answer in the location where the offer was located. Set the location as the media descriptor or session level.
Default: 60
Example: set mikey-time-tolerance 90
symmetric-address-failure: Secondary property. Specifies whether the system learns the source IP address from the RTP/RTCP packets, even if the packets fail decryption. When enabled, the first packet in a particular stream that fails SRTP decryption causes a DroppedPacket notification to be sent to the application with the address of the packet. The application treats this like a srcIPChanged notification.
Default: disabled
Values: enabled | disabled
Example: set symmetric-address-failure enabled
treat-as-secure: Secondary property. Specifies whether a proprietary security indicator is used on the SIP interface; either ST-secure or ST-insecure. This setting only operates on the X-Siemends-Call-Type header when MIKEY encryption is involved in pass-thru mode.
Default: disabled
Values: disabled: Sets ST-insecure
enabled: Sets ST-secure
auto: If SRTP is active on both sides of the call, the ME allows the X-Siemends-Call-Type header to pass unchanged. If SRTP is not active on other side of the call, the header is set to ST-insecure. The ”auto” value sets the interface to trusted.
Example: set treat-as-secure enabled
This object determines the ToS value setting for the in-leg of the session. This ToS value determines the quality of service that the call receives. The ME marks the ToS field of all packets it sends out on the in-leg with the value you specify. Enter a number that represents the 8-bit Differentiated Services (DS) field of the IP packet in decimal format, such as 26 for 00011010 or 104 for 01101000. This value can be of use to upstream devices.
Configures the SDP hold attributes that are sent to an endpoint. The in-hold-translation object applies to SDP bodies sent to the endpoint that initiated the call. The out-hold-translation object applies to SDP bodies sent to the endpoint that initially received the call. When a call/stream is put on hold, the endpoint putting the call on hold sends an SDP offer with certain recognizable SDP characteristics: SDP connection information (c-line) and hold attributes that typically include ”inactive” or ”sendonly.” The endpoint that is being put on hold responds with an SDP answer acknowledging the hold. This is normally expressed by including a ”recvonly” or ”inactive” hold attribute. Not all servers recognize all SDP hold characteristics, so these objects can be used to configure the SDP hold characteristics sent to a given server. The ME recognizes the following hold attributes (for offer or answer): sendrecv, sendonly, recvonly, and inactive. When changing or removing hold attributes, the ME will remove or overwrite any of these attributes in the SDP.
For example, an endpoint being put on hold may responds to a server offer with ”a=inactive” or ”a=recvonly.” Some servers may interpret an SDP answer of ”a=inactive” as ”I'm not listening, do not send me music-on-hold.” If the endpoint putting the call on hold will play music-on-hold anyway, you may configure the ME change the answer attribute to ”a=recvonly.” In that way, when the server receives the SDP answer with the ”a=recvonly,” it may still play the music-on-hold.
config vsp default-session-config in-hold-translation config vsp policies session-policies policy name rule name session-config in-hold-translation config vsp dial-plan dial-prefix entryName session-config in-hold-translation config vsp dial-plan route name session-config in-hold-translation config vsp dial-plan source-route name session-config in-hold-translation config vsp session-config-pool entry name in-hold-translation
admin: Specifies whether this hold translation entry is applied to calls matching the session configuration.
Default: disabled
Values: enabled disabled
Example: set admin enabled
offer-address: Specifies how the ME modifies the SDP connection information (c-line) sent in an SDP offer.
Default: pass
Values: pass: Not modify the c-line.
zero: Change the address reported in the c-line to 0.0.0.0.
non-zero: Change the address reported in the c-line to the last known address, typically the system interface address.
Example: set offer-address zero
offer-attribute: Specifies how the ME modifies the SDP hold attributes in the SDP offer.
Default: pass
Values: pass: Not modify the hold attributes.
remove: Remove any recognized hold attributes from SDP.
inactive: Set the hold attribute to ”inactive.”
sendonly: Set the hold attribute to ”sendonly.”
sendrecv: Set the hold attribute to ”sendrecv.” Use with care; this setting effectively tells the endpoint that the stream is not on hold.
Example: set offer-attribute inactive
answer-address: Specifies how the ME modifies the SDP connection information (c-line) sent in an SDP answer.
Default: pass
Values: pass: Not modify the c-line.
zero: Change the address reported in the c-line to 0.0.0.0.
non-zero: Change the address reported in the c-line to the last known address, typically the system interface address.
Example: set answer-address zero
answer-attribute: Specifies how the ME modifies the SDP hold attributes in the SDP answer.
Default: pass
Values: pass: Not modify the hold attributes.
remove: Remove any recognized hold attributes from SDP.
inactive: Set the hold attribute to ”inactive.”
sendonly: Set the hold attribute to ”sendonly.”
sendrecv: Set the hold attribute to ”sendrecv.” Use with care; this setting effectively tells the endpoint that the stream is not on hold.
Example: set answer-attribute remove
remove-telephone-events: Secondary property. Specifies whether the system strips telephone-events from the SDP when a call is placed on hold. When set to true, the system does strip events, which may be necessary for some phones (Polycom, for example). When false, the system does not modify events in the SDP.
Default: false
Values: true | false
Example: set remove-telephone-events true
Changes the media descriptor string (e.g., the CODEC for audio or video) in the SDP. Use this in cases where a client is unable to understand a variation in name of a CODEC/media descriptor. For example, G729 is sometimes transmitted as G729a. Inbound media normalization applies to the segment from the initiator to the ME device.
config vsp default-session-config in-media-normalization config vsp policies session-policies policy name rule name session-config in-media-normalization config vsp dial-plan dial-prefix entryName session-config in-media-normalization config vsp dial-plan route name session-config in-media-normalization config vsp dial-plan source-route name session-config in-media-normalization config vsp session-config-pool entry name in-media-normalization
normalize: Specifies, for a media type, how to normalize a CODEC/media descriptor name. The initial subtype is the type the system matches on and replaces. The alternate subtype is the type that the system then inserts in the SDP to replace the initial subtype. You can select a pre-configured type or enter a custom type.
Default: audio Values: <audio | video | application | image | custom-mime-type mimeType>[initialSubType] [alternateSubType]
Example: set normalize video g729 g729a
Configure in-media-scanner-settings. When in-media-scanner-settings are configured, a media scaner is started on the in-leg of the call and reports events based on the analysis of the received audio from the endpoint.
config vsp default session-config in-media-scanner-settings config vsp session-config-pool entry <name> in-media-scanner-settings config vsp policies session-policies policy <name> rule <name> session-config in-media-scanner-settings
admin: Enables or disables the in-media scanner settings.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
pre-scan-time: The number of milliseconds to delay before invoking the in-media scanner. This property is not applicable for the on-demand media scanner.
Default: 20
Values: Min: 0 / Max: 4294967295
low-threshold: Enter the talk or stable tone signal power threshold in dbs. Crossing this threshold indicates quiet.
Default: -36
Values: Min: -36 / Max: 3
Example: set low-threshold -25
high-threshold: Enter the quiet signal power threshold in dbs. Crossing this threshold indicates talking or stable tone.
Default: -36
Values: Min: -36 / Max: 3
Example: set high-threshold -25
low-long-duration: The number of milliseconds of detected quiet before reporting a long-pause, otherwise a short pause is reported.
Default: 2000
Values: Min: 0 / Max: 4294967295
Example: set low-long-duration 1500
high-long-duration: The number of milliseconds of detected talk or tone before reporting a long talk or stable tone, otherwise a short talk is reported.
Default: 900
Values: Min: 0 / Max: 4294967295
Example: set high-long-duration 1500
averaging-window: Secondary property The amount of time in dbs used when calculating signal strength.
Default: 100
Values: Min: 10 / Max: 1000
Example: set averaging-window 500
nominal-rounding-factor: Secondary property. The signal strength is rounded to the nearest multiple of this value before comparing against other signal strengths.
Default: 2
Values: Min: 1 / Max: 25
Example: set nominal-rounding-factor 4
event-report-frequency: The number of milliseconds the media scanner should wait between generation of media scanner events. Setting this property to 0 causes the media scanner events to be reported immediately as they occur.
Default: 1000
Values: Min: 0 / Max: 60000
Example: set event-report-frequency 39
event-report-count-threshold: The maximum number of media scanner events that can be pended for waiting for the event-report-frequency timer to expire before being reported. If the number of queued media scanner events reaches this count, all of the events will be immediately reported.
Default: 25
Values: Min: 1 / Max: 6000
Example: set event-report-count-threshold 73
event-report-flags: Set the media scanner events to report flags.
Default: report all events
Values: short-pause, long-pause, short-talk, long-talk, stable-tone
Example: set event-report-flags short-pause
Configures in-leg Message Session Relay Protocol (MSRP) interworking.
config default-session-config in-msrp-session-leg
config session-config-pool entry <name> in-msrp-session-leg
admin: Enable or disable MSRP interworking on this call leg.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
msrp-leg-transport: Specify the MSRP transport method for RCS or WebRTC.
Default: TCP
Values: TCP | TLS | WS | WSS
Example: set msrp-leg-transport TLS
connection-reuse: Not currently supported.
default-media-interface: Specify the local media interface to use for an MSRP connection if svc-routing fails to locate the appropriate interface.
Default: There is no default setting
Example: set default-media-interface int1
use-mdesc-cline-first: (Advanced) Specify whether the MSRP session manager attempts to use the SDP c-line (before using the path attribute) to learn the remote MSRP endpoint's media IP address.
Default: false
Values: true | false
Example: set use-mdesc-cline-first false
socket-read-size: (Advanced) Specify the MSRP socket read size to use when assembling incoming MSRP messages.
Default: 4096
Min: 0 / Max: 4294967296
Example: set socket-read-size 5000
partial-forward-size: (Advanced) Specify the threshold for forwarding buffered MSRP message content bytes.
Default: 1024
Min: 0 / Max: 4294967296
Example: set partial-forward-size 2050
connsrc-match-path: (Advanced) Specify whether the ME allows incoming MSRP connections even when a remote address does not match the SDP path attribute.
Default: false
Values: true | false
Example: set connsrc-match-path false
allow-missing-fingerprint: (Advanced) Specify whether the ME allows an MSRP secure connection even when the SDP fingerprint attribute is missing.
Default: false Values: true | false
Example: set allow-missing-fingerprint false
Customizes call admission control, on a per-AOR basis, for inbound calls only. This is in contrast to the admission control settings found in the location-call-admission-control object, which sets aggregate inbound and outbound limits. Use the show location-cac status provider to view call admission control settings and counters for an AOR.
config vsp default-session-config location-call-admission-control inbound-controls config vsp policies session-policies policy name rule name session-config location-call-admission-control inbound-controls config vsp dial-plan dial-prefix entryName session-config location-call-admission-control inbound-controls config vsp dial-plan route name session-config location-call-admission-control inbound-controls config vsp dial-plan source-route name session-config location-call-admission-control inbound-controls config vsp session-config-pool entry name location-call-admission-control inbound-controls
max-number-of-concurrent-calls: Specifies the maximum number of active incoming and outgoing calls allowed for this AOR at one time. When this value is reached, the connection does not accept calls until the value drops below the threshold.
A value of 0 causes the system to decline all calls and registrations.
Default: 1000
Values: Min: 0 / Max: 1000000
Example: set max-number-of-concurrent calls 1500
max-calls-in-setup: Sets the maximum number of simultaneous inbound and outbound call legs in setup stage that are allowed for this AOR. A call leg in setup is much more compute-intensive than established call legs, so this value is more restrictive than the concurrent call leg value. A value of 0 causes the system to decline all calls and registrations.
Default: 30
Values: Min: 0 / Max: 10000
max-bandwidth: Secondary property. Specifies the amount of bandwidth the system allocates to the AOR. When the system reaches the maximum bandwidth limit for a server, it rejects calls until bandwidth use drops below the maximum.
Default: unlimited Values: unlimited | kbps
Example: set max-bandwidth 512
call-rate-limiting: Secondary property. Limits the number of calls sent to the AOR within a certain interval. Once this interval is reached, the system rejects any calls to this AOR, returning a response code and message, until the rate decreases. This feature sets the acceptable arrival rate for incoming calls.
If enabled, set the number of calls allowed and the measurement interval (in seconds). You can also enter a result code from 400 to 699 and a text string to accompany call rejection if no available server is found.
Default: disabled; if set to enabled, the default calls-per-interval is 60, the default interval is 1, and the default result is 486, Busy Here Values: enabled [calls-per-interval][interval][result-code][result-string] | disabled
Example: set call-rate-limiting enabled 50 1 480 ”Temporarily unavailable”
The inbound-header-settings configuration object allows you to set fields to remove and/or replace header settings in the SIP headers for inbound traffic.
config vsp policies session-policies policy default rule session-config inbound-header-settings
pAssert-mode: Secondary property. Sets whether or not to strip the number in the P-Asserted-Identity field from the SIP header. When enabled, the ME replaces the value in the From field with the value from the P-Asserted-Identity field for the outbound call leg. (Note that the ME maintains the original From field value in the Contact field.)
Default: disabled
Values: enabled | disabled
Example: set pAssert-mode enabled
header-to-strip: Secondary property. Configures the ME to strip the value of the specified field. Enter a SIP header field name.
Default: There is no default setting
Example: set header-to-strip sip1
allowed-header: Sets the SIP headers that should be explicitly allowed to remain in the SIP message. You can enter any number of header names by re-executing the command.
Default: There is no default-setting
Example: set allowed-header header1
blocked-header: Sets the SIP headers that should be explicitly removed from the SIP message. You can enter any number of header names by re-executing the command.
Example: set blocked-header header5
apply-allow-block-to: Sets whether the allow and block properties of this object apply to request messages, response messages, or both.
Default: requests-and-responses
Values: requests: Apply to requests only
responses: Apply to responses only
requests-and-responses: Apply to requests and responses
Example: set apply-allow-block-to responses apply-to-allow-block-to-dialog: Specifies whether the allow and block properties of this object apply to a specific dialog or not.
Default: both
Values: inbound: Apply to the inbound dialog only
outbound: Apply to the outbound dialog only
both: Apply to both inbound and outbound dialogs
Example: set apply-to-allow-block-to-dialog inbound
sip-manipulation: Specify the configured sip-manipulation you want to associate with this header-setting. Configure the sip-manipulation in the sip-manipulation-pool > sip-manipulation object.
Default: There is no default setting
Example: set sip-manipulation sipmanip1
Specifies whether the ME modifies the content of the host, port, and/or transport fields of the REQUEST URI. If set, changes are applied only to the REQUEST message traveling in the opposite direction of the session initiation REQUEST message. (Use the request-uri-specification object to change outbound REQUEST URIs.) These properties should only be changed to override the default behavior because of issues with an intermediary device.
config vsp default-session-config inbound-request-uri-specification config vsp policies session-policies policy name rule name session-config inbound-request-uri-specification config vsp dial-plan dial-prefix entryName session-config inbound-request-uri-specification config vsp dial-plan route name session-config inbound-request-uri-specification config vsp dial-plan source-route name session-config inbound-request-uri-specification config vsp session-config-pool entry name inbound-request-uri-specification
host-use-next-hop: Specifies whether to change the host portion of the REQUEST URI. If enabled, the system sets the host portion to the IP address of the next hop. If disabled, the host portion remains unchanged.
Default: disabled
Values: enabled | disabled
Example: set host-use-next-hop enabled
port-use-next-hop: Specifies whether to change the port specified in the REQUEST URI. If enabled, the system sets the port number to the port used for the next hop. If disabled, the port number remains unchanged.
Default: disabled
Values: enabled | disabled
Example: set port-use-next-hop enabled
transport-use-next-hop: Specifies whether to change the transport protocol specified in the REQUEST URI. If enabled, the system sets the transport to the protocol used by the next hop. If disabled, the transport protocol remains unchanged.
Default: disabled
Values: enabled | disabled
Example: set transport-use-next-hop enabled
Configures events for incoming SIP messages. This is a secondary object.
config vsp session-config-pool entry name event-settings inbound-sip-messages
config vsp default-session-config event-settings inbound-sip-messages
admin: Enables or disables events for incoming SIP messages.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
apply-to-methods-for-events: Select the SIP methods you want the ME to create events for.
Default: There is no default setting
Example: set apply-to-methods-for-events INVITE
apply-to-responses: Specifies whether to send events for SIP requests or both requests and responses. When no, changes are applied to requests only. When yes, changes are applied to requests and responses. If yes, you must set the response code to which it applies. A value of 0 implies all responses.
Default: no Values: no | yes <response code>
Example: set apply-to-responses yes 0
apply-to-dialog: Specifies whether to send events for SIP messages on a specific dialog or not.
Default: both
Values: inbound: Apply to the inbound dialog only
outbound: Apply to the outbound dialog only
both: Apply to both the inbound and outbound dialogs.
Example: set apply-to-dialog inbound
cseq: Secondary property. Sets a mechanism to further filter for which SIP messages have events sent. When set to 0 (the default), the ME sends events for all SIP messages. If set to any other value, the ME only sends events for SIP messages having a CSEQ field that matches that value.
Default: 0
Values: Min: 0 / Max: 4294967296
Example: set cseq 100
Enables IM archiving, applies text stamps before or after IM messages, and sends IM message alerts to the configured the ME event log.
config vsp default-session-config instant-messaging config vsp policies session-policies policy name rule name session-config instant-messaging config vsp dial-plan dial-prefix entryName session-config instant-messaging config vsp dial-plan route name session-config instant-messaging config vsp dial-plan source-route name session-config instant-messaging config vsp session-config-pool entry name instant-messaging
directive: Assigns an action to the message.
Note that if you specify the refuse directive with text, the text is placed on the method line of the SIP response message. That line is usually not displayed to the user. If you want a message displayed to the sender, use the message-to-sender property.
Default: follow-sip-directive
Values: allow: Allows the message, even if higher-level policy (under instant-messaging) says to refuse or discard it.
discard: Silently deletes the message instead of delivering it. No notification is sent.
refuse <resultCode><resultString>: Deletes the message, but sends a SIP error response to the sending agent. Optionally, specify the result code, between 400 and 699, and/or a result string to send in the error response. The default error code is 400, with no accompanying text.
follow-sip-directive: Follows whatever actions are configured at the session-level. (These are the settings under sip-directive, and/or instant-messaging.)
Example: set directive refuse 500 Message discarded”
alert: Sends alert messages containing IM session information and message content to the configured system event log. Specify the configured event logging target path and severity level.
Default: disabled Values: enabled logTargetReference severity | disabled
Example: set alert enabled ”services event-log file messages” info
archiving: Enables or disables archiving of SIP instant messages to the system database. When enabled, the system records to its database all instant messages (the text in the body of MESSAGE SIP messages) of sessions matching the policy. Messages are recorded in both directions.
Note that you must enable the database-write property of the vsp object for archiving to work.
You can view the instant messages that have been archived using the ME Management System Call Logs feature.
Default: disabled
Values: enabled | disabled
Example: set archiving enabled
pre-stamp: Prepends the user-specified text before the IM message content in this SIP session.
Default: There is no default setting
Example: set pre-stamp ”Good Morning”
post-stamp: Appends the specified test after the IM message content in this SIP session.
Default: There is no default setting
Example: set post-stamp ”Have a great day”
message-to-sender: Sets the text message to send back to the originating IM sender in this SIP session.
Default: There is no default setting
Example: set message-to-sender ”Messages to this user are logged.”
message-to-recipient: Sets the text message to send to the IM recipient in this SIP message. This text is in addition to the incoming message.
Default: There is no default setting
Example: set message-to-recipient ”Message is being logged”
Creates pointers to configured word lists and/or URL lists. For more detailed information on instant message filtering, see Configuring IM Filtering Objects. To create word lists, see the word-list object; to create URL lists, see the url-list object.
config vsp default-session-config instant-messaging-content config vsp policies session-policies policy name rule name session-config instant-messaging-content config vsp dial-plan dial-prefix entryName session-config instant-messaging-content config vsp dial-plan route name session-config instant-messaging-content config vsp dial-plan source-route name session-config instant-messaging-content config vsp session-config-pool entry name instant-messaging-content
word-list: Configures a pointer to a previously configured word list. You can include any number of words lists in your instant messaging content scan.
Default: There is no default setting
Example: set word-list ”vsp im-filtering word-list bad-words”
url-list: Configures a pointer to a previously configured URL list. You can include any number of URL lists in your instant messaging content scan.
Default: There is no default setting
Example: set url-list ”vsp im-filtering url-list good-guys”
Customizes call admission control on a per-AOR basis. These parameters can also be configured in the location service settings object. From that object, the values are applied across the VSP. The settings override the VSP-wide settings and are applied when an AOR registers. The parameters remain in effect until the next registration period. Use the show location-cac status provider to view call admission control settings and counters for an AOR.
config vsp default-session-config location-call-admission-control config vsp policies session-policies policy name rule name session-config location-call-admission-control config vsp dial-plan dial-prefix entryName session-config location-call-admission-control config vsp dial-plan route name session-config location-call-admission-control config vsp dial-plan source-route name session-config location-call-admission-control config vsp session-config-pool entry name location-call-admission-control
max-number-of-concurrent-calls: Specifies the maximum number of active incoming and outgoing calls allowed for this AOR at one time. When this value is reached, the connection does not accept calls until the value drops below the threshold.
A value of 0 causes the ME to decline all calls and registrations.
Default: 1000
Values: Min: 0 / Max: 1000000
Example: set max-number-of-concurrent calls 1500
max-calls-in-setup: Sets the maximum number of simultaneous inbound and outbound call legs in setup stage that are allowed for this AOR. A call leg in setup is much more compute-intensive than established call legs, so this value is more restrictive than the concurrent call leg value. A value of 0 causes the system to decline all calls and registrations.
Default: 30
Values: Min: 0 / Max: 10000
Example: set max-calls-in-setup 50
admission-control: Specifies whether the system considers AOR limitations when forwarding a call from the AOR. The system tracks the number of concurrent (both incoming and outgoing) active calls for this AOR. If this property is enabled, the system does not forward calls from the AOR if the limit has been reached and instead sends a ”603 Declined” message. If disabled, the system does forward calls from the AOR. (Set the call limit with the max-number-of-concurrent-calls property.) See Admission Control for an AOR for additional information.
Default: disabled
Values: enabled | disabled
Example: set admission-control enabled
max-bandwidth: Secondary property. Specifies the amount of bandwidth the system allocates to the AOR. When the system reaches the maximum bandwidth limit for a server, it rejects calls until bandwidth use drops below the maximum.
Default: unlimited Values: unlimited | kbps
Example: set max-bandwidth 512
emission-control: Specifies whether the system considers AOR limitations when forwarding a call to this AOR. The system tracks the number of concurrent (both incoming and outgoing) active calls for the AOR. If this property is enabled, the system does not forward calls to the AOR if the limit, set with the max-number-of-concurrent-calls property, has been reached. Instead, the system sends one of the following messages and drops the call:
If there is one outbound server/UAC/UAS, the system sends a ”486 Busy” message, indicating that the route was resolved but that the AOR was unavailable.
If there are multiple outbound server/UAC/UASs and all have reached the maximum concurrent calls threshold, the system sends a ”486 Busy” message.
If there are multiple outbound server/UAC/UASs and at least one has not reached the maximum concurrent calls threshold, the return code is determined by the final server that the system attempted to reach. This could be, for example, ”486 busy” or a ”504 server timeout” if the last server was unresponsive and the transaction timed out.
If disabled, the system continues to forward calls to the AOR. See Admission Control for an AOR for more information.
Default: disabled
Values: enabled | disabled
Example: set emission-control enabled
call-rate-limiting: Specifies whether the system considers AOR limitations when forwarding a call to this AOR. The system tracks the number of concurrent (both incoming and outgoing) active calls for the AOR. If this property is enabled, the system does not forward calls to the AOR if the limit, set with the max-number-of-concurrent-calls property, has been reached. Instead, the system sends one of the following messages and drops the call:
If there is one outbound server/UAC/UAS, the system sends a ”486 Busy” message, indicating that the route was resolved but that the AOR was unavailable.
If there are multiple outbound server/UAC/UASs and all have reached the maximum concurrent calls threshold, the system sends a ”486 Busy” message.
If there are multiple outbound server/UAC/UASs and at least one has not reached the maximum concurrent calls threshold, the return code is determined by the final server that the system attempted to reach. This could be, for example, ”486 busy” or a ”504 server timeout” if the last server was unresponsive and the transaction timed out.
If disabled, the system continues to forward calls to the AOR. See Admission Control for an AOR for more information.
Default: disabled; if set to enabled, the default calls-per-interval is 60, the default interval is 1, and the default result is 486, Busy Here Values: enabled [calls-per-interval][interval][result-code][result-string] | disabled
Example: set call-rate-limiting enabled 50 1 480 ”Temporarily unavailable”
Sets the method by which the system implements bridged line appearance (BLA) for a phone. BLA is a feature that allows a phone to display line appearance (status) on more than one phone. A single telephone number (or SIP URL) can be monitored by more than one user agent, and when a call is made to the number, all subscribed user agents can respond. The status of the call is then displayed on all phones mapped to the number. This object sets the mechanism used to do that mapping, either the default SIP mechanism, or the method put forth in the IETF draft, Implementing Bridged Line Appearances Using Session Initiation Protocol (SIP).
config vsp default-session-config location-events config vsp policies session-policies policy name rule name session-config location-events config vsp dial-plan dial-prefix entryName session-config location-events config vsp dial-plan route name session-config location-events config vsp dial-plan source-route name session-config location-events config vsp session-config-pool entry name location-events
bridged-line-appearance: Sets the type of bridged line appearance the phone uses. Select default for most phones. Select draft-anil-sipping-bla for the following configurations:
When using a Sylantro phone
When using sticky trunk ports, where the registration-plan > route > alter-contact property is set to trunk-port-per-aor, -endpoint, or -binding
Default: default
Values: default | draft-anil-sipping-bla
Example: set bridged-line-appearance draft-anil-sipping-bla
Customizes the manner in which the ME executes a location lookup. Within this object you can specify which headers the ME uses to perform the lookup, and in what order the fields should be searched on. Use this, for example, to send INVITEs to an AOR registered on the ME but without using the AOR of this device in the Request URI.
config vsp default-session-config location-lookup config vsp policies session-policies policy name rule name session-config location-lookup config vsp dial-plan dial-prefix entryName session-config location-lookup config vsp dial-plan route name session-config location-lookup config vsp dial-plan source-route name session-config location-lookup config vsp session-config-pool entry name location-lookup
sequence: Selects the header field(s) used to do the lookup in the location cache. Enter as many headers as needed by re-executing the command.
Default: There is no default setting
Values: request-uri | to-uri | from-uri | contact | trunk-port
Example: set sequence request-uri
match-uri-params: Specifies which URI parameters should be used in the location lookup comparison. By default, the system does a base comparison of the received contact using the following URI information:
scheme (i.e., sip, sips, tel)
user
host
port (5060 if not otherwise specified)
transport (UDP if not otherwise specified)
maddr
Enter a string for the ME Engine to match against.
Default: There is no default setting
Example: set match-uri-params state
Normalizes the Request and/or To URI in a REQUEST message. The ME replaces the URI portion of the specified headers with the AOR data found in the location cache. With the properties of this object you can update the To and Request headers with the full AOR or just the host portion from the AOR in the location cache. This configuration provides a means of normalizing requests destined for an AOR (e.g., phone).
config vsp default-session-config location-normalization config vsp policies session-policies policy name rule name session-config location-normalization config vsp dial-plan dial-prefix entryName session-config location-normalization config vsp dial-plan route name session-config location-normalization config vsp dial-plan source-route name session-config location-normalization config vsp session-config-pool entry name location-normalization
request-uri: Specifies the portion of the AOR to use in normalizing the Request URI. You can replace the AOR portion of the Request URI with the entire AOR from the location cache or the domain name only (AOR-host-only). By default, no replacement occurs.
Default: none
Values: none | AOR | AOR-host-only
Example: set request-uri AOR
to-uri: Specifies the portion of the AOR to use in normalizing the To URI. You can replace the AOR portion of the Request URI with the entire AOR from the location cache or the domain name only (AOR-host-only). By default, no replacement occurs.
Default: none
Values: none | AOR | AOR-host-only
Example: set to-uri AOR
Enables or disables session logging on a per-call basis. Because session logging can result in large amounts of data, this object allows you to enable logging only on calls that match certain criteria. Note that logging must also be enabled in the event-log object for the session details to be written to a target.
config vsp default-session-config log-alert config vsp policies session-policies policy name rule name session-config log-alert config vsp dial-plan dial-prefix entryName session-config log-alert config vsp dial-plan route name session-config log-alert config vsp dial-plan source-route name session-config log-alert config vsp session-config-pool entry name log-alert
message-logging: Enables or disables logging of individual SIP messages (INVITE, CANCEL, BYE, etc.) to the system database. Select no-registers to log all but SIP REGISTER messages. (This can also be accomplished with the master-services database object sip-register property, but message-logging is the preferred method.). Message logging can only be enabled if you have the appropriate license for the feature.
The system uses the database records to:
Display the call detail diagrams at the ME Management System and logs which show the individual SIP messages involved in each session.
Provide collected messages for detection of attack patterns, which are then used to construct DOS rules blocking the attack traffic.
Default: enabled if you have the license for message logging features and disabled if you do not
Values: enabled: Logging is active
disabled: Logging is inactive
no-reregisters: Logging is active for all but REGISTER messages/sessions
invite-session-only: Logging is active for only INVITE-based sessions
filtered: Logging is filtered by apply-to-methods-for-filtered logs property.
Example: set message-logging enabled
apply-to-methods-for-filtered-logs: Specifies the type of SIP message to be logged if message-logging is set to filtered. If set to any other option, this property is ignored.
Default: INVITE
Example: set apply-to-methods-for-filtered-logs INVITE+REFER
alert: Secondary property. Enables or disables the sending of session alert messages to the configured system logging target. If set to enabled, specify the path to a previously configured logging target and a severity level. The path must be specified in quotation marks.
Default: disabled Values: enabled logTargetReference severity | disabled
Example: set alert enabled ”services event-log file messages” info
logging: Secondary property. Enables or disables event logging for this session. If session logging is enabled, and the event-log object enables logging, details are recorded in a target file.If session logging is disabled, even if the event-log is enabled, the system does not write session events to the log.
Default: disabled
Values: enabled | disabled
Example: set logging enabled
tracing: Secondary property. Enables SIP-related tracing for the session. When enabled, you can exit to a SIP shell and enter the trace-filter command to see related traces. (Note that you must have advanced CLI permissions to execute shell commands.)
Default: disabled
Values: enabled | disabled
Example: set tracing enabled
message-auditing: Secondary property. Enables the system to maintain an audit trail of changes to each SIP message.
Default: disabled; if enabled, the default severity level is error Values: enabled severity | disabled
Example: set message-auditing enabled error
Configures the SIP media anchoring settings to apply to this SIP call session.
Note:
You must define and enable a pool of ports specifically for media services. Without these ports defined, the ME cannot establish the anchoring necessary to provide media services. See the media-ports object for more information.Real-time Control Protocol (RTCP) is a companion protocol to RTP that gathers statistics on the performance and quality of the SIP call connection. When enabled, RTP monitors the quality of the SIP call and conveys information about the SIP call session. It is based on the periodic transmission of control packets to all participants in the session, and provides feedback on the quality of the data distribution.
RTCP statistics are used to dynamically adjust and optimize the call quality for current network conditions. You can configure the ME to drop, pass, and generate RTCP packets and to record the statistical data in its database through the media object.
The ME provides a session maintenance feature for use in cases of abnormal session terminations. Normally, SIP signaling causes the creation and termination of each media session. In the event of network or device failure, however, terminating SIP messages may not be received by the signaling system, or the signaling system may be unable to request that the resources be released by the media proxy. You can enable the inactivity-timeout property to recover resources for aborted media sessions.
When inactivity-timeout is enabled, the media proxy periodically checks for inactive media sessions. If a session timed out due to inactivity, a message is sent to the signaling the ME device, which logs the event and sends the appropriate SIP signaling messages to notify each party of the call. The media proxy then releases the resources for the inactive session.
The ME supports transcoding media types, which is the process of converting media from one CODEC into a different CODEC on output. This allows, in some cases, endpoints supporting different media types to communicate. You add CODECs using the transcode-media-types property, augmenting the list of CODECs contained in an INVITE SDP offer. Note that the order in which a CODEC appears in the offer/answer matters in the ME selection. Original media types appear first, followed by the media types added in this object. You can re-order the media types with the move command, or using the in-codec-preferences or out-codec-preferences objects.
The following table illustrates and explains the transcoding process.
Table 39-3 Transcoding Media Types
| Device | CODEC |
|---|---|
|
Phone East |
Supports A, B, Z |
|
Phone West |
Supports B, C, Z |
|
ME |
Configured with transcoding support for C |
|
ME |
Does not support Z |
As a result, the following sequence occurs:
EAST's original INVITE/SDP offering contains CODECs A, B, Z.
The ME augments the list with CODEC C, resulting in an offer of A, B, Z, C in the INVITE/SDP.
WEST's original OK/SDP response answers with CODECs B, C, Z.
The ME modifies and forwards the response to EAST with CODECs B, Z.
The following table shows how the ME behaves with this configuration:
Table 39-4 ME Actions and Reasons
| ME Action | Reason |
|---|---|
|
Forwards Z packets in both directions unchanged. |
Because CODEC Z is unsupported. |
|
Transcodes C packets from WEST to B on their way to EAST. |
Because B is the first supported CODEC in the response (on behalf of WEST) to EAST. That response is comprised of the WEST response minus CODECs that the ME added. (Note that if there are no remaining CODECs after this process, the ME adds back in the first CODEC common to EAST and the system.) |
|
Forwards B packets from either direction, no transcoding necessary. |
Because both sides understand B. |
Use the rtp-transcode-stats and rtp-transcode-summary status providers to view active and summary statistics for RTP transcoding.
config vsp default-session-config media config vsp policies session-policies policy name rule name session-config media config vsp dial-plan dial-prefix entryName session-config media config vsp dial-plan route name session-config media config vsp dial-plan source-route name session-config media config vsp session-config-pool entry name media
anchor: Enables or disables SIP media session anchoring on this SIP call session. Media anchoring forces the SIP media session to traverse the system. The auto setting enables conditional anchoring. In this case, the system uses its auto-anchoring algorithms to determine anchoring necessity based on a variety of criteria, including whether you have configured smart anchoring via the autonomous-ip object and whether the calling devices are behind a firewall.
Default: enabled
Values: enabled | disabled
Example: set anchor disabled
transcode-media-types: Specifies the CODEC(s) that the system can use for transcoding media. Adding CODECs through this property augments the list contained in an INVITEs SDP offer. See Transcoding Media Types for a full description and example.
You can enter any number of valid input CODEC types. To see a list of available input CODECs, type a question mark at the command line.
Default: There is no default setting
Example: set transcode-media-types g726-16
auto-conference: Allows a user to log into a conference call automatically by prepending the assigned username and passcode before the number to be dialed. To do this, the SIP session establishes the initial call, and then identified DTMF tone strings are used to join the conference, creating the conference codes.
Use the pre- and post-tone-delay properties to allow announcements to play from the conference site before the digits are entered. Use the VSP dtmf-generation object to set parameters for the conference codes created.
Default: disabled, with no regular expression specified and the outbound side hearing the tones
Values: administrative state: Turns on or off the auto-conference feature. When enabled, the system strips the Request URI and plays the first portion of the regular expression as tones.
regular expression: Identifies the digit strings in the user portion of the Request URI. All but the last item matched are played as tones after the call comes up. The system replaces the last match as the outgoing user portion in the Request URI.
direction: Specifies which side of the call hears the tone string.
For example, if the Request URI is 1234567#123#johnd@webex.com, the resulting digit strings played are 1234567# and 123#. The Request URI becomes johnd@webex.com.
Example: set auto-conference enabled (.*#) (.*#) (.*) out
pre-tone-delays: Specifies the milliseconds of delay prior to playing the DTMF tone string that was determined from the auto-conference property. You can specify multiple delays to correspond to multiple matches of the regular expression. If the number of matches exceeds the number of delay entries, the last delay entry is repeated. For example, if you have three matches in the regular expression, and have configured delays of three and five milliseconds, the system delays 3 milliseconds, plays tone 1, delays 5 milliseconds, plays tone 2, delays 5 milliseconds, plays tone 3.
Default: There is no default setting
Example: set pre-tone-delays 3
post-tone-delays: Specifies the milliseconds of delay following the playing of a DTMF tone string that was determined from the auto-conference property. You can specify multiple delays to correspond to multiple matches of the regular expression. If the number of matches exceeds the number of delay entries, the last delay entry is repeated. For example, if you have three matches in the regular expression, and have configured delays of three and five milliseconds, the system plays tone 1, delays 3 milliseconds, plays tone 2, delays 5 milliseconds, plays tone 3, delays 5 milliseconds.
Default: There is no default setting
Example: set post-tone-delays 3
introduction: Specifies the path to a WAV file that plays at the introduction of a call (no audio is sent through until the introduction completes). Use the file-play-verify action to ensure that the recording is of a format supported by the ME device.
Default: There is no default setting
Example: set introduction /cxc_common/intro1.wav
music-on-hold: Specifies the path to a WAV file that plays (in a loop) while the call is on hold. Use the file-play-verify action to ensure that the recording is of a format supported by the ME device.
Default: There is no default setting
Example: set music-on-hold /cxc_common/hold1.wav
inactivity-timeout: Specifies whether the system can timeout an anchored media session due to inactivity. See Media Session Maintenance for more information. If you enable this feature, you must set the length of the inactivity timer. See Setting Time and Time Intervals for information on entry format requirements.
Default: disabled; if set to enabled, the default timer setting is 3600 Values: enabled seconds (greater than 60) | disabled
Example: set inactivity-timeout enabled 1800
inactivity-style: Specifies which parties of a call must stop sending RTP before the system activates the inactivity timer (if enabled). When set to session, the system activates the timer when all parties stop sending RTP. When set to per-call-leg, if one party stops sending RTP, the system activates the inactivity timer.
Default: session
Values: session | per-call-leg
Example: set inactivity-style per-call-leg
monitor: Associates a playback configuration with the call session. The playback function allows you to record SIP calls for playback on the ME Management System or a configured endpoint. Enter a pointer to a previously configured monitor-group object.
Default: There is no default setting
Example: set playback ”vsp monitor-group callRecord”
packet-marking: Enables or disables packet marking. Marking (tagging) a packet provides a quality of service (QOS) indicator, which routers along the path may act on. You could use packet marking, for example, to give priority to voice calls over other traffic. The system writes the value you enter to the TOS (or DiffServ) field of the IP header.
Default: 0xa0 Values: disabled | tos value (0-255)
Example: set packet-marking tos 128
rtp-stats: Enables or disables the collection and logging of RTP and call quality statistics to the system database. Note that this property must be enabled to:
Display Mean Opinion Score (MOS) or Quality of Service (QoS) statistics for a call at the ME Management System.
Display RTP values in accounting files or databases that the system is writing to.
Default: disabled
Values: enabled | disabled
Example: set rtp-stats enabled
rtcp <action><true | false>: Specifies the handing and generation of RTCP packets in this SIP call session. When configuring this property, you set an action for call senders and a logging capability. Note that this property is not available when performing transcoding. See the Release Notes for more information.
Set an action that defines how the system responds to RTCP packets it receives from call senders.
Default: pass false
Values: pass: Transmits all packets.
drop: Drops any packets.
generate-only-if-required: Generates RTCP packets if it detects that the sender is not generating them.
generate-always: Always generates packets, regardless of whether the sender did.
Configure whether to log session statistics to the system database:
true: System writes RTCP statistics to the database, along with its own statistics.
false: System ignores statistics.
Example: set rtcp generate-only-if-required false
mirror: Specifies whether calls that match the defined policy are mirrored to other boxes in the cluster. To use this feature you must also set mirror-media-settings to true in the cluster object.
Default: enabled
Values: enabled | disabled
Example: set mirror disabled
answer-media-loopback: Sets whether the system answers a loopback call. When enabled, the system answers the call and generates RTP according to the negotiations. When disabled, the system allows the call to proceed according to the rest of the configuration. Endpoints can use these loopback calls to test the quality of the media transport, in accordance with the IETF draft-ietf-mmusic-media-loopback-07.txt.
Default: disabled
Values: enabled | disabled
Example: set answer-media-loopback enabled
tag-routing: Specifies whether tag routing is in use for media. If routing tags are configured for an interface (using the ip > routing-tag property), and that interface has media configured on it, the tags are only used when this property is enabled. See Tag-Based Route Selection for more information.
Default: enabled
Values: enabled | disabled
Example: set tag-routing disabled
encode-auto-anchor-tag: Secondary property. Specifies whether the content of the x-cxc-info field of the SDP is encoded or in clear text. This property is only applicable if the anchor property is set to auto. The x-cxc-info field contains the information necessary for the system to make an auto anchoring decision. If set to true, the field is base-64 encoded. If set to false it is sent in clear text.
Default: true
Values: true | false
Example: set encode-auto-anchor-tag false
transcode-balance-ptime: Secondary property. Specifies whether the system uses signaling to attempt to ”coax” the originating phone to send RTP packets at the rate of the destination phone. When transcoding, RTP packets may be arriving at departing at different rates (as determined by the CODEC in use with the phone). When true, the system sends the originating phone the ptime value (interval) in use by the destination phone.
Default: true
Values: true | false
Example: set transcode-balance-ptime false
transcode-auto-release: Secondary property. Specifies whether the system passes packets without transcoding, thereby releasing the transcode license for those sessions. This only applies if the system is set to auto anchor (with the anchor property set to auto) and transcode (using transcode-media-types property) and auto anchoring is required due to reachability issues between the source and destination. When this property is set to true in that situation, if the source and destination have the same set of CODECs, then the system passes the packets without transcoding. If false, the system transcodes the packets, costing the license two sessions for the duration.
Default: true
Values: true | false
Example: set transcode-auto-release false
decode-telephone-events: Secondary property. Specifies whether the system should decode DTMF packets and inject them into the audio stream. This property may be used in cases where only one of the endpoints supports DTMF per RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. If false, the default, the phone is responsible for inserting the DTMF into the audio stream. If true, the system decodes the DTMF packets and inserts the resulting audio into the stream.
Default: false
Values: true | false
Example: set decode-telephone-events true
repair-empty-codec-list: Secondary property. Specifies whether the system repairs an SDP that contains an illegal media description. When set to true, the system inserts a default well-known CODEC into the SDP, allowing the phone to process the message. It also sets the port to 0 (if it is not already), disabling the media stream. When set to false, the system makes no changes to the SDP.
This property only applies to illegal video and audio media types. The system inserts payload type 34 (H263) for video and payload type 0 (PCMU) for audio as the default payload types.
Default: false
Values: true | false
Example: set repair-empty-codec-list true
strip-blocked-stream: Secondary property. Specifies whether the system removes the m= line from the SDP when all CODECs within that line are blocked. CODECs can be blocked by several mechanisms: media-type filtering, CODEC preferences, and stripping of unplayable or unverifiable CODECs. If the system blocks all CODECs in an m= line, it disables the stream. If this property is set to true, the system removes the entire m= line from the SDP. If set to false, the stream is disabled (by setting the port to zero) but remains in the SDP with some CODECs. This property is only applicable in messages that contain multiple media streams (e.g., audio and video).
Default: false
Values: true | false
Example: set strip-blocked-stream true
preserve-sdp-order: Secondary property. Sets whether the system attempts to preserve the SDP attribute order. When set to true, the system makes a best attempt to preserve the order. When false, the default, the system uses the order native to its SDP parser.
Default: false
Values: true | false
Example: set preserve-sdp-order true
handle-unknown-lines-in-sdp: Secondary property. Specifies how the system handles errored lines that it receives in an SDP. If set to strip, the default, the system removes the offending lines and forwards the packet. If set to pass, the system forwards the packet untouched. If set to error, the system sends a 488 message (Not Acceptable) back to the sender and logs a message to the event log.
Default: strip
Values: strip | pass | error
Example: set handle-unknown-lines-in-sdp error
rtp-min-consecutive: Secondary property. Specifies the number of consecutive RTP packets the system must receive to establish an RTP source as valid.
Default: 3
Example: set rtp-min-consecutive 5
rtp-sequence-discontinuity: Secondary property. Specifies whether the system monitors for, detects, and corrects RTP sequence number discontinuity. In some cases, a gateway may change the CODEC for a packet, but keep the same synchronization source (SSRC). If the resulting sequence numbers are discontinuous, it causes problems for SRTP processing. When this property is enabled, the system changes the SSRC if it detects sequence problems. When disabled, it does nothing.
Default: disabled
Values: enabled | disabled
Example: set rtp-sequence-discontinuity enabled
rtp-splice: Secondary property. Configures a mechanism for maintaining RTP parameters. When enabled, the ME attempts to maintain RTP parameters when it injects DTMF (or other RTP audio) into the RTP stream. This may result in additional processing on each RTP packet after the normal audio stream is resumed, but it is required for interoperability with some endpoints because they do not recognize the DTMF when RTP parameters change (SSRC, sequence numbers, and timestamps). Note that the rtp-stats property must be enabled for this property to work.
Default: disabled
Values: enabled | disabled
Example: set rtp-splice enabled
combine-recording-fragments: Secondary property. Specifies whether the system checks the state of the RTP recording file when a call ends. If enabled, when an ME-recorded call ends, the media master determines whether the RTP recording file is fragmented across the cluster. (Fragmenting can occur when a failover causes part of the recording to reside on one box and part on another.) If there are fragments, the system copies each to the master box and assembles the entire file. When disabled, the system bypasses fragment checking, which boosts performance.
Default: enabled
Values: enabled | disabled
Example: set combine-recording-fragments disabled
auto-anchor-consider-nat: Secondary property. Specifies whether to disable a portion of the ME anchoring algorithm. This property is only applicable if you have the anchor property set to auto. Typically, if the system is forwarding a call from behind a NAT, it would anchor the media stream. You may set this property to disabled if, for example, you have a phone behind a NAT destined for a server that can do NAT traversal and you want to release the media stream.
Default: enabled
Values: enabled | disabled
Example: set auto-anchor-consider-nat disabled
default-session-bandwidth: Secondary property. Specifies an initial bandwidth value to use when calculating the bandwidth usage for each leg of a media session. The resulting value (in accumulation with all other session values) determines whether a server pool server has reached the configured max-bandwidth setting.
Note that the bandwidth usage value is based not on the actual traffic on the wire, but on a calculation done by the ME device. The calculation uses the value associated with the first known CODEC identified in the SDP for a usage rate. If there is not a known CODEC, or the value has not yet been determined from the SDP, the system uses this setting.
If you set this property to zero, the system treats media streams with no known codecs in the SDP as using zero bandwidth. (They do not count against the bandwidth limit for a given server.)
Default: 87
Values: Min: 0 / Max: 10000
Example: set default-session-bandwidth 31
SIP-response-code-on-media-resource-alloc-failures: Secondary property. Specifies the SIP response code to send when there is a media allocation failure. The failure may occur for one of two reasons: either media ports are not configured on the interface or no ports are available because all configured media ports are in use.
Default: 488 (Service Unavailable)
Values: Min: 300 / Max: 699
Example: set SIP-response-code-on-media-resource-alloc-failures
attributeless-auto-anchor: Secondary property. When enabled in conjunction with the anchor-mode=auto, the ME attempts to auto-anchor streams without additional ME attributes in the SDP.
Default: disabled
Values: enabled | disabled
Example: set attributeless-auto-anchor enabled
release-provisionally-anchored-media: Release media resources that have been provisionally anchored.
When media > anchor is set to auto, the ME attempts to determine when anchoring resources can be released based on IP addresses and routing-tags of the communicating endpoints. When these determinations cannot be made, the media is deemed ”provisionally anchored.” In releases previous to Release 3.5.5, provisionally allocated media was released by default.
Default: false
Values: true | false
Example: set release-provisionally-anchored-media false
report-last-timestamp: When enabled, the ME reports the timestamp of the last received media packet.
Default: disabled
Values: enabled | disabled
Example: set report-last-timestamp enabled
monitor-rfc-2833: Specifies whether to have the ME change SSRC when it detects RTP sequence number discontinuity on active SSRC.
Default: disabled
Values: enabled | disabled
Example: set monitor-rfc-2833 enabled
discard-media-on-hold: Secondary property. Specifies whether to discard media from an endpoint when that endpoint is placed on hold.
Default: disabled
Values: enabled | disabled
Example: set discard-media-on-hold enabled
pass-candidate-attributes: Specifies whether or not ICE candidate SDP attributes received by the ME are forwarded to an endpoint.
Default: disabled
Values: enabled | disabled
Example: set pass-candidate-attributes enabled
propagate-reinvite-from-header: When enabled, when the ME receives an Invite request, the ME switches to the new From: header when it is different in a reinvite. When disabled, the ME uses the From: header received in the initial Invite.
Default: disabled
Values: enabled | disabled
Example: set propagate-reinvite-from-header enabled
dtmf-detected-events: Specifies whether received DTMF events are reported to web services.
Default: disabled
Values: disabled: Provides the normal DTMF handling based on the session-config > dtmf-preferences settings or the older legacy dtmf-translation settings.
report-and-forward: DTMF events are reported to web services and translated and forwarded based on the session-config > dtmf-preference settings or the older legacy dtmf-translation settings.
report-and-discard: DTMF events are reported to web services and then discarded without being translated.
Example: set dtmf-detected-events report-and-discard
Configure media scanner settings. When media-scanner-settings are enabled, the media-scanner is started after the outgoing call connects and the pre-scan-time has elapsed. The media-scanner monitors the signal strength and duration of the received audio to divide it into intervals.
config vsp default session-config media-scanner-settings config vsp session-config-pool entry <name> media-scanner-settings config vsp policies session-policies policy <name> rule <name> session-config media-scanner-settings
admin: Enables or disables the media scanner settings for play-file-broadcast.
Default: disabled
Values: enabled | disabled
Example: set admin enabled
pre-scan-time: The number of milliseconds to delay before invoking the media scanner for speaker detection.
Default: 20
Values: Min: 0 / Max: 4294967295
Example: set pre-scan-time 35
max-scan-time: The maximum number of milliseconds before canceling media scanning due to timeout.
Default: 30000
Values: Min: 0 / Max: 4294967295
Example: set max-scan-time 25000
low-threshold: Enter the quiet signal power threshold in dbs.
Default: -36
Values: Min: -36 / Max: 3
Example: set low-threshold -25
high-threshold: Enter the talk or tone signal power threshold in dbs.
Default: -36
Values: Min: -36 / Max: 3
Example: set high-threshold -25
low-long-duration: The number of milliseconds of detected quiet before declaring a long-pause.
Default: 2000
Values: Min: 0 / Max: 4294967295
Example: set low-long-duration 1500
high-long-duration: The number of milliseconds of detected talk or tone before declaring a long-talk or stable-tone.
Default: 900
Values: Min: 0 / Max: 4294967295
Example: set high-long-duration 1500
averaging-window: Secondary property The window of time used when calculating signal strength.
Default: 100
Values: Min: 10 / Max: 1000
Example: set averaging-window 500
nominal-rounding-factor: Secondary property. The signal strength is rounded to the nearest multiple of the value you enter for this property.
Default: 2
Values: Min: 1 / Max: 25
Example: set nominal-rounding-factor 4
Sets the media types that are allowed and/or prohibited during the session. You select a media type: audio, video, application, or MIME: and then a specific subtype. Use the question mark character at the command line to see a list of available subtypes. For example:
config media-type> set allowed-media-types audio ?
allow sessions to use these media types
syntax: set allowed-media-types audio sub-type
set allowed-media-types video sub-type
set allowed-media-types application sub-type
set allowed-media-types custom-mime-type mime-type sub-type
any
pcmu
gsm
g723
dvi4
lpc
pcma
g722
--more--
In addition to the pre-configured options, you can allow or block custom types that may be part of your enterprise.
config vsp default-session-config media-type config vsp policies session-policies policy name rule name session-config media-type config vsp dial-plan dial-prefix entryName session-config media-type config vsp dial-plan route name session-config media-type config vsp dial-plan source-route name session-config media-type config vsp session-config-pool entry name media-type
allow-media-types: Sets the media types allowed during the session. Re-execute the command for each type you want to allow.
Default: The default subtype setting for audio, video, and application is any; there is no default setting for custom-mime-type. Values: audio subType | video subType | application subType | custom-mime-type mimeType subType
Example: set allowed-media-types custom-mime-type application safe-trade
blocked-media-types: Sets the media types to prohibit during the session. Re-execute the command for each type you want to block.
Default: The default subtype setting for audio, video, and application is any; there is no default setting for custom-mime-type. Values: audio subType | video subType | application subType | custom-mime-type mimeType subType
Example: set blocked-media-types video mpls
Enables or disables media verification for RTP sessions, and sets SIP session termination based on RTP and alert messaging. The media-verify-config object allows you to verify RTP and RTCP media streams negotiated over SIP sessions. The settings verify that the media traffic passing through the ME matches the negotiated and legal scheme for CODECs (coder/decoders) operating on SIP signals.
admin: Enables or disables the current media-verify-config object. If enabled, the system uses these settings if this object is included in the session configuration media object.
Default: enabled Values: enabled | disabled
Example: set admin disabled
terminate-session: Enables or disables SIP session termination if a RTP media verification alert is generated. If enabled, the system terminates a media session completely upon the first media verification error (B2B mode only).
Default: enabled
Values: enabled | disabled
Example: set terminate-session disabled
alert-frequency: Sets the number of milliseconds the system should wait between generation of media alert messages, which indicate a media verification error.
Default: 1000
Values: Min: 10 / Max: 10000
Example: set alert-frequency 2000
Configures multimedia stream-specific settings when configuring the multimedia stream server (MSS) process on the ME.
config vsp session-config-pool entry name multimedia-stream-settings
config vsp default-session-config multimedia-stream-settings
stream-name: Specify how the stream name is derived. You can enter a string or keep the default value, session-id.
Default: session-id
Example: set set-stream-name session-id2
in-leg-permissions: Sets the stream's read and write permissions for the in-leg.
Default: read-write
Values: read-only: The stream is read-only.
write-only: The stream is write-only.
read-write: The stream is read and write enabled.
Example: set in-leg-permissions read-only
out-leg-permissions: Sets the stream's read and write permissions for the out-leg.
Default: read-write
Values: read-only: The stream is read-only.
write-only: The stream is write-only.
read-write: The stream is read and write enabled.
Example: set out-leg-permissions read-only
to-header-specification: Select the To: header provided to the streaming code.
Default: to-header
Example: set to-header-specification to-header
invert-uri-index: When enabled, the ME swaps the publish and subscribe indices in the RTMP URIs.
Default: disabled
Values: enabled | disabled
Example: set invert-uri-index enabled