With the introduction of multi-tenancy support on CUBE, the sip-specific attributes can be configured at per tenant basis
in addition to the existing global or dial-peer levels.
The
voice class tenant
<tag> command allows sip-specific attributes
to be configured at per tenant basis. The command
voice class tenant
<tag> can be then applied to individual
dial-peers, thereby associating them to a particular tenant. See the following
table "Multi-tenant
Configuration List" for information on the complete list of
configurations present under the
voice class tenant
<tag> .
If tenants are
configured under dial-peer, then configurations are applied in the following
order of preference.
-
Dial-peer
configuration
-
Tenant
configuration
-
Global
configuration
That is, if the
value of the attribute under dial-peer configuration is system, then the value
is taken from the tenant configuration. And, if the value under the tenant
configuration is also system, then the global configuration is used.
If there are no
tenants configured under dial-peer, then the configurations are applied using
the default behavior in the following order:
-
Dial-peer
configuration
-
Global
configuration
The following table
lists the various configurations present under
voice class tenant
<tag> . For more information on specific
configurations, see the
Voice and Video command
reference guide lists.
Note
|
Attributes that
are not available under
voice class tenant
<tag> use the default behavior—With
preference of dial-peer followed by the global configuration.
|
Table 1. Multi-Tenant
Configuration List
Command
|
Description
|
aaa
|
SIP-UA AAA
related configuration
|
anat
|
Allow
alternative network address types IPv4 and IPv6
|
asserted-id
|
Configure
SIP UA privacy identity settings
|
associate
|
Associate
a RCB for outgoing calls
|
asymmetric
|
Configure
global SIP asymmetric payload support
|
authentication
|
Digest
Authentication Configuration
|
bandwidth
|
Allow SIP
SDP bandwidth-related options
|
bind
|
SIP bind
command
|
block
|
Block 18X
response to INVITE
|
call-route
|
Configure
call routing options
|
conn-reuse
|
Reuse the
sip registration tcp connection for the end-point behind a Firewall
|
connection-reuse
|
Use
listener port for sending requests over UDP
|
contact-passing
|
302
contact to be passed through for CFWD
|
content
|
Content
carried as part of SIP message
|
copy-list
|
Configure
list of entities to be sent to peer leg
|
credentials
|
User
credentials for registration
|
disable-early-media
|
Disable
early-media cut through
|
dns
-a-override
|
Skip DNS
A/AAAA query when SRV query timesout
|
dscp
-profile
|
DSCP
Profile global config
|
early-media
|
Configure
method to handle early-media Update Request
|
early-offer
|
Configure
sending Early-Offer
|
encap
|
Configure
SDP encapsulation
|
error-code-override
|
Configure
sip error code
|
error-
passthru
|
SIP error
response pass-thru functionality
|
exit
|
Exits
from the voice class configuration mode
|
g729
|
G729
codec interoperability settings
|
handle-replaces
|
Handle
INVITE with REPLACES header at SIP spi
|
header-passing
|
SIP Headers need to be passed to applications
|
help
|
Description of the interactive help system
|
history-info
|
History
Info header support
|
host-registrar
|
Use
sip-ua registrar value in Diversion and Contact header for 3xx messages
|
interop-handling
|
Enable
interop-handling
|
localhost
|
Specify
the DNS name for the localhost
|
map
|
Mapping
options
|
max-forwards
|
Change
number of max-forwards for SIP Methods
|
midcall
-signaling
|
Configure method to handle mid-call signaling
|
nat
|
SIP nat
global config
|
no
|
Negate
a command or set its defaults
|
notify
|
SIP
Signaling Notify Configuration
|
offer
|
Configure settings for Offers made from the Gateway
|
options-ping
|
Send
OPTION pings to remote end
|
outbound-proxy
|
Configure an Outbound Proxy Server
|
pass-thru
|
SIP
pass-through global config
|
permit
|
Permit
hostname for this gateway
|
preloaded-route
|
Use
pre-loaded route header for outgoing calls, if available
|
privacy
|
Configure SIP UA privacy settings
|
privacy-policy
|
Set
privacy behavior for outgoing SIP messages
|
random-contact
|
Use
Random Contact for outgoing calls, if available
|
random-request- uri
|
Configure options for Request-URI having random value
|
reason-header
|
Configure settings for supporting SIP Reason Header
|
redirection
|
Enable
call redirection (3xx) handling
|
refer-
ood
|
Configure maximum number of out-of-dialog refer made to the Gateway
|
referto
-passing
|
Refer-To needs to be passed through for transfer
|
registrar
|
Configure SIP registrar VoIP Interface
|
registration
|
Enable
registration options
|
rel1xx
|
Type of
reliable provisional response support
|
remote-party-id
|
Enable
Remote-Party-ID support in SIP User Agent
|
requri
-passing
|
Request
URI needs to be passed through
|
reset
|
SIP
Reset Options
|
retry
|
Change
default retries for each SIP Method
|
send
|
Configure outgoing message options
|
session
|
SIP
Voice Protocol session config
|
sip-profiles
|
SIP
Profiles global config
|
sip-server
|
Configure a SIP Server Interface
|
srtp
|
Allow
SIP related SRTP options
|
srtp-auth
|
Allow
to set preferred suites
|
tel-config
|
Tel
format cfg for headers other than req -line in
|
timers
|
SIP
Signaling Timers Configuration
|
update-
callerid
|
Enable
sending updates for callerid
|
url
|
Url
configuration for request-line url in outgoing INVITE
|
video
|
Video
related config for sip
|
warn-header
|
SIP
Warning-Header global config
|
xfer
|
Transfer target configuration
|