The following
table describes the properties that you can set to configure the SIP Service.
The first time you configure the SIP service on a Call Server, you must restart
the Call Server.
Configuration
Enable Outbound
Proxy
Select
Yes to use a use a Cisco Unified SIP proxy server.
For more information on configuring the Cisco Unified SIP Proxy Server, consult
the CUSP documentation.
Default
|
Range
|
Restart Required
|
No
|
Yes and No
|
Yes
|
Use DNS SRV type
query
Select
Yes to use DNS SRV for outbound proxy lookup.
Otherwise, select
No. See
Load-Balancing SIP Calls
for information on using DNS SRV for load-balancing SIP calls.
 Note |
If you enable
Resolve SRV records locally, you must select
Yes to ensure the feature works properly.
|
Default
|
Range
|
Restart Required
|
No
|
Yes and No
|
Yes
|
Resolve SRV records locally
Select to resolve
the SRV domain name with a local configuration file instead of a DNS Server.
 Note |
If you enable
Resolve SRV records locally, you must select
Yes to use DNS SRV type query. Otherwise, this
feature will not work.
|
See the
Configuration
Guide for Cisco Unified Custom Voice Portal for additional information
about local SRV configuration.
Default
|
Range
|
Restart Required
|
None
|
Enabled or Disabled No
|
Yes
|
Outbound Proxy Host
If you selected
Enable Outbound Proxy, select an Outbound Proxy Server from the drop-down list.
These are the SIP Proxy Servers that have been added to the Operations Console.
For information on configuring a SIP Outbound Proxy Server, consult the CUSP
documentation.
Default
|
Range
|
Restart Required
|
No
|
Valid IP Address
|
Yes
|
Outbound SRV domain
name/Server group name (FQDN)
If you use a
hostname that is an SRV type record instead of a standard DNS type record, this
field contains a fully qualified domain name that is configured on the DNS
server. Otherwise, the field contains an SRV configuration file.
For example,
outbound calls made from CVP SIP service will be addressed to the URL of
sip:<label>@<srvfqdn>. Redundant proxy servers, for example, can
route calls using such a configuration.
Default
|
Range
|
Restart Required
|
None
|
Follows
the same validation rules as hostname, which includes uppercase and lowercase
letters in the alphabet, the numbers 0 through 9, and a dash.
0 - 256
character length.
|
Yes
|
DN on the Gateway to Play
the Ringtone
Dialed Number (DN)
configured on the gateway to play ringtone (dedicated VoIP dial peer).
To learn the DN
configured on the gateway to play ringtone, execute the
sh
run
command on the gateway and look for the dial peer that matches the
incoming dialed number. See
Ringtone Dialed Number Learning on Gateway Example.
Default
|
Range
|
Restart Required
|
9191
|
Any valid label
|
No
|
DN on the Gateway to Play
the Error Tone
Dialed Number (DN)
configured on the gateway to play the error.wav file (dedicated VoIP dial
peer).
To learn the DN
configured on the gateway to play the error tone, execute the
sh
run
command on the gateway and look for the dial peer that matches the
incoming dialed number. See
Ringtone Dialed Number Learning on Gateway Example.
Default
|
Range
|
Restart Required
|
9292
|
Any valid label
|
No
|
Override System Dialed
Number Pattern Configuration
Use the new Dialed
Number Pattern system configuration, but maintain the existing Call Server
interface.
Default
|
Range
|
Restart Required
|
Unchecked
|
The
override check box's default state differs depending on the device state:
-
For
new devices, override is disabled (unchecked). New Unified CVP Call Server
devices will use configured system-level dialed number patterns by default.
-
For
upgraded devices, override is enabled (checked). Upgraded Unified CVP Call
Server devices will use device-level dialed number patterns by default.
|
No
|
Advanced
Configuration
Outbound proxy
port
Specify the port
to be used.
Default
|
Range
|
Restart Required
|
5060
|
|
No
|
Outgoing transport
type
Specifies the
outgoing transport, you can set it as TCP or UDP.
Default
|
Range
|
Restart Required
|
TCP
|
TCP or UDP
|
Yes
|
Port number for incoming
SIP requests
Specify the port
to be used foe incoming SIP requests.
Default
|
Range
|
Restart Required
|
5060
|
|
Yes
|
Incoming transport
type
Specifies the
incoming transport type.
Default
|
Range
|
Restart Required
|
TCP+UDP
|
TCP, UDP, TCP+UDP
|
Yes
|
Time to wait for ICM
instructions
Specifies the wait
time in milliseconds for ICM instructions. It is optional value for the list
addition.
Default
|
Range
|
Restart Required
|
2000
|
|
No
|
SIP info tone
duration
Specifies the wait
time in milliseconds for SIP info tone. It is optional value for the list
addition.
Default
|
Range
|
Restart Required
|
100
|
|
No
|
SIP info comma
duration
Specifies the wait
time in milliseconds for SIP info comma. It is optional value for the list
addition.
Default
|
Range
|
Restart Required
|
100
|
|
Yes
|
Generic Type Descriptor
(GTD) parameter Forwarding
To be added
Default
|
Range
|
Restart Required
|
UUS
|
|
No
|
Prepend digits
Specifies the
number of digits to be removed for SIP URI user number.
Default
|
Range
|
Restart Required
|
0
|
0-20
|
No
|
UDP Retransmission
Count
Specifies the
number of UDP retransmission will be attempted.
Default
|
Range
|
Restart Required
|
3
|
|
No
|
Use Error Refer
Flag for play
error tone when call fails to caller.
Default
|
Range
|
Restart Required
|
False
|
True or False
|
Yes
|
IOS Gateway Options Dynamic
Routing
Default
|
Range
|
Restart Required
|
|
True or False
|
Yes
|
IOS Gateway Options
Reporting
Reports on
resource utilization.
Default
|
Range
|
Restart Required
|
|
True or False
|
Yes
|
QoS
Select QoS Level
Select the Quality of Service (QoS) level between the SIP Service and the SIP Proxy Server.
 Note |
For more information, see the Enterprise QoS Solution Reference Network Design Guide.
|
Default |
Range |
Restart Required |
cs3 |
The drop-down list has the following values: af11, af12, af13, af21, af22, af23, af31, af32, af33, af41, af42, af43, cs1,
cs2, cs3, cs4, cs5, cs6, cs7,default, ef
|
Yes |
Security
Properties
Incoming secure
port
Specify the port
to be used.
Default
|
Range
|
Restart Required
|
5061
|
|
No
|
Incoming secure
protocol
This option is
grayed out as it is prepopulated.
Default
|
Range
|
Restart Required
|
TLS
|
|
No
|
Outgoing secure
protocol
This option is
grayed out as it is prepopulated.
Default
|
Range
|
Restart Required
|
TLS
|
|
No
|
Supported TLS
Versions
This allows to
select the versions of TLS to be supported for securing the SIP signaling on
the IVR leg. The TLS versions currently supported are TLSv1.0, TLSv1.1, and
TLSv1.2.
Default
|
Range
|
Restart Required
|
TLS v1.2
|
TLSv1.0, TLSv1.1, and TLSv1.2
|
Yes
|
 Note |
When you select
a a given TLS version, Unified CVP supports SIP TLS requests for that version
and the higher supported versions.
|
Supported
Ciphers
This field defines
the ciphers, which is supported by Unified CVP, with key size lesser than or
equal to 1024 bits.
The default cipher
is TLS_RSA_WITH_AES_128_CBC_SHA, which is pre-populated and cannot be deleted
as it is mandatory for TLSv1.2.
Cipher configuration is available only if TLS is enabled.
Default
|
Range
|
Restart Required
|
TLS_RSA_WITH_AES_128_CBC_SHA
|
|
Yes
|
 Note |
If you are using CUBE version 16.6 and higher, you must manually change the crypto suite to 128 by enabling CLI on the dial-peer
towards CVP as shown:
voice class srtp-crypto 1
crypto 1 AES_CM_128_HMAC_SHA1_32
dial-peer voice xxxx voip (Dial-peer to CVP)
...
voice-class sip srtp-crypto 1
|
SIP Header
Passing (to ICM)
Header Name
Specify the SIP
header name and click
Add to add it to the list of SIP headers passed to
ICM.
Default
|
Range
|
Restart Required
|
None
|
Maximum length of 210 characters. |
No
|
Parameter
This field is
optional for list addition.
Default
|
Range
|
Restart Required
|
None
|
Maximum length of 210 characters. |
No
|
Local Static
Routes
 Note |
Enable "Override
System Dialed Number Pattern Configuration" to configure these values.
|
Dialed Number (DN)
Creates a Static
Proxy Route Configuration Table. You must create static routes if you do not
use a SIP Proxy Server. Before adding a local static route, you must enter a
value into both the Dialed Number (DN) and IP Address fields so that the local
static route is complete.
Click
Add to create a proxy route using the Dialed Number
(DN) and the IP address/Hostname entered above the
Add button. The newly created proxy route is added
to the list of proxy routes displayed in the box below the Add button.
Click
Remove to delete the selected DN from the list box
of Dialed Numbers.
Default
|
Range
|
Restart Required
|
None
|
Dialed number pattern, destination (must be format of
NNN.NNN.NNN.NNN or a hostname). See
Valid Formats for Dialed Numbers
for more information.
|
No
|
IP Address/Hostname/Server
Group Name
The IP address,
hostname, or server group SRV domain name.
 Note |
If you use
Server Group Name, you must select
Yes to use
DNS SRV type
query and you must enable
Resolve SRV records locally to ensure the feature
works properly.
|
Default
|
Range
|
Restart Required
|
None
|
Valid IP address, hostname, or SRV domain name
|
No
|
Dialed Number
(DN) Patterns
 Note |
Enable "Override
System Dialed Number Pattern Configuration" to configure these values.
|
Patterns for sending calls
to the originator :
Dialed Number
(DN)
Creates a SIP Send
Back to Originator Lookup Table. Specify the DN patterns to match for sending
the call back to the originating gateway for VXML treatment. For the Unified
CVP branch model, use this field to automatically route incoming calls to the
Call Server from the gateway back to the originating gateway at the branch. For
information on the Unified CVP branch model, see
Planning
Guide for Cisco Unified Customer Voice Portal.
This setting
overrides sending the call to the outbound proxy or to any locally configured
static routes. It is also limited to calls from the IOS gateway SIP "User
Agent" because it checks the incoming invite's User Agent header value to
verify this information. If the label returned from ICM for the transfer
matches one of the patterns specified in this field, the call is routed to
sip:<label>@<host portion of from header of incoming
invite>.
Three types of DNs
work with Send To Originator: VRU label returned from ICM, Agent label returned
from ICM, and Ringtone label.
Send To
Originator does not work for the error message DN because the inbound error
message is played by survivability and the post-route error message is a SIP
REFER. (Send To Originator does not work for REFER transfers).
 Note |
For Send To
Originator to work properly, the call must be TDM originated and have
survivability configured on the pots dial peer.
|
Patterns for RNA timeout on
outbound SIP calls:
- Dialed Number (DN)
Creates a Dialed
Number (DN) pattern outbound invite timeout using the DN and Timeout entered
above the Add button. Click
Add to add the newly created DN pattern outbound
invite timeout to the list displayed in the box below the Add button.
Click
Remove to delete the selected DN pattern outbound
invite timeout from the list.
Timeout (Seconds)
The number of
seconds the SIP Service waits for transferee to answer the phone or accept the
call.
If a selected
termination (for either a new or transferred call) returns a connection failure
or busy status, or if the target rings for a period of time that exceeds the
Call Server's ring-no-answer (RNA) timeout setting, the Call Server cancels the
transfer request and sends a transfer failure indication to Unified ICM. This
scenario causes a router requery operation. The Unified ICM routing script then
recovers control and has the opportunity to select a different target or take
other remedial action.
Default
|
Range
|
Restart Required
|
60 seconds
|
5 - 60
|
No
|
Custom ringtone
patterns:
Dialed Number (DN)
Specify a custom
Dialed Number (DN) pattern. Click
Add to add the newly created DN pattern to the list
displayed in the box below the Add button.
Ringtone Media file name
The file name of
the ringtone to be played for the respective dialed number.
The ringtone media
file must be saved to the VXML Gateway. See
Transfer Script and Media File to Gateway
for more information.
Default
|
Range
|
Restart Required
|
None
|
0 - 256
characters. Spaces are not permitted.
Provide
the URL for the stream name in the following form:
rtsp://<streaming server IP address>
/<port>/<foldername>/ <filename>.rm
|
No
|
Post Call
Survey DNIS Mapping
 Note |
Enable "Override
System Dialed Number Pattern Configuration" to configure these values.
|
Incoming Call Dialed Number
(DN)
Click
Add to add the newly created DN pattern to the list
displayed in the box below the Add button. Click
Remove to delete the selected DN pattern from the
list.
Default
|
Range
|
Restart Required
|
None
|
Dialed Number pattern, destination (must be in the form of
NNN.NNN.NNN.NNN or a hostname). See
Valid Formats for Dialed Numbers
for more information.
|
No
|
Survey Dialed Number
(DN)
Click
Add to add the newly created DN to the list
displayed in the box below the Add button. Click
Remove to delete the selected DN from the list.
Default
|
Range
|
Restart Required
|
None
|
Accepts only alphanumeric characters
|
No
|