Step 1
| Log in as the Provider, Reseller, or Customer Administrator.
|
Step 2
| Make sure
the hierarchy path is set to the node where the
Cisco Unified
Communications Manager
is configured.
|
Step 3
| Perform one of
- If you logged in as the
Provider or Reseller Administrator, select
Device
Management > CUCM > SIP Trunks.
- If you logged in as the
Customer Administrator, select
Device
Management > Advanced > SIP Trunks.
|
Step 4
| Perform one of
- To add a new SIP trunk,
click
Add,
then go to Step 4.
- To edit an existing SIP
trunk, choose the SIP trunk to be updated by clicking on its box in the
leftmost column, then click
Modify to edit the selected SIP trunk. Go to Step 5.
|
Step 5
| From the
CUCM
pulldown menu, select the hostname, domain name, or IP address of the
Cisco Unified
Communications Manager
to which you want to add the SIP trunk.
Note
|
The
CUCM
pulldown menu only appears when a SIP trunk is added; it does not appear when
you edit a SIP trunk.
|
Important:
The only
Cisco Unified
Communications Managers
that appear in the
CUCM
pulldown list are
Cisco Unified
Communications Managers
that are located
at the
node where you are adding the SIP trunk, and
all
Cisco Unified
Communications Managers
in hierarchies above the node where you are adding the SIP trunk. To provision
a
Cisco Unified
Communications Manager
server, refer to the “Installation Tasks” section of
Installing Cisco Unified Communications Manager.
|
Step 6
| Enter a unique
name for the new SIP trunk in the
Device
Name field, or modify the existing
Device Name
if desired.
|
Step 7
| From the
Device
Information tab, modify the following fields as required.
Option |
Description |
Device Name
(Mandatory)
|
Enter a unique identifier for the trunk using up to 50
alphanumeric characters: A-Z, a-z, numbers, hyphens (-) and underscores (_)
only.
Default
value: None
|
Trunk Service Type
(Mandatory)
|
Select
one of
- None—Choose this option
if the trunk is not used for call control discovery, Extension Mobility Cross
Cluster, or Cisco Intercompany Media Engine
-
Call
Control Discovery—Choose this option to enable the trunk to support call
control discovery.
-
Extension Mobility Cross Cluster—Choose this option to enable
the trunk to support the Extension Mobility Cross Cluster (EMCC) feature.
Choosing this option causes the following settings to remain blank or unchecked
and become unavailable for configuration, thus retaining their default values:
Media Termination Point Required, Unattended Port, Destination Address,
Destination Address IPv6, and Destination Address is an SRV.
-
Intercompany Media Engine—Ensure that the Cisco IME server is
installed and available before you configure this field.
-
IP
Multimedia Subsystem Service Control (ISC)—Choose this option to enable the
trunk to support IP multimedia subsystem service control.
Default
value: None (Default)
|
Description
(Optional) |
Enter a
descriptive name for the trunk using up to 114 characters in any language, but
not including double-quotes ("), percentage sign (%), ampersand (&),
backslash (\), or angle brackets (<>).
Default
value: empty
|
Device Pool
|
Choose
the appropriate device pool for the trunk. For trunks, device pools specify a
list of
Cisco Unified
Communications Managers
that the trunk uses to distribute the call load dynamically.
Note
|
Calls
that are initiated from a phone that is registered to a
Cisco Unified
Communications Manager
that does not belong to the device pool of the trunk use different
Cisco Unified
Communications Managers
of this device pool for different outgoing calls. Selection of
Cisco Unified
Communications Manager
nodes occurs in a random order. A call that is initiated from a phone that is
registered to a
Cisco Unified
Communications Manager
that does belong to the device pool of the trunk uses the same
Cisco Unified
Communications Manager
node for outgoing calls if the
Cisco Unified
Communications Manager
is up and running.
|
Default
value: Default
|
Common Device
Configuration
(Optional)
|
Choose
the common device configuration to which you want this trunk assigned. The
common device configuration includes the attributes (services or features) that
are associated with a particular user.
Default
value: None
|
Call Classification
(Mandatory)
|
This
parameter determines whether an incoming call through this trunk is considered
off the network (OffNet) or on the network (OnNet). When the Call
Classification field is configured as Use System Default, the setting of the
Cisco Unified
Communications Manager
clusterwide service parameter, Call Classification, determines whether the
trunk is OnNet or OffNet. This field provides an OnNet or OffNet alerting tone
when the call is OnNet or OffNet, respectively.
Default
value: Use System Default
|
Media Resource Group List
(Optional)
|
This list provides a prioritized grouping of media resource
groups. An application chooses the required media resource, such as a Music On
Hold server, from among the available media resources according to the priority
order that a Media Resource Group List defines.
Default
value: None
|
Location
(Mandatory)
|
Use
locations to implement call admission control (CAC) in a centralized
call-processing system. CAC enables you to regulate audio quality and video
availability by limiting the amount of bandwidth that is available for audio
and video calls over links between locations. The location specifies the total
bandwidth that is available for calls to and from this location.
Select
the appropriate location for this trunk:
-
Hub_None—Specifies that the locations feature does not keep
track of the bandwidth that this trunk consumes.
-
Phantom—Specifies a location that enables successful CAC across
intercluster trunks that use H.323 protocol or SIP.
-
Shadow—Specifies a location for intercluster enhanced location
CAC. Valid for SIP intercluster trunks (ICT) only.
Default
value: Hub_None
|
AAR Group
(Optional)
|
Choose the automated alternate routing (AAR) group for this
device. The AAR group provides the prefix digits that are used to route calls
that are otherwise blocked due to insufficient bandwidth. An AAR group setting
of None specifies that no rerouting of blocked calls is attempted.
Default
value: None
|
Tunneled Protocol
|
Select the QSIG option if you want to use SIP trunks or SIP
gateways to transport (tunnel) QSIG messages from
Cisco Unified
Communications Manager
to other PINXs. QSIG tunneling supports the following features: Call Back, Call
Completion, Call Diversion, Call Transfer, Identification Services, Path
Replacement, and Message Waiting Indication (MWI).
Note
| Remote-Party-ID (RPID) headers coming in from the SIP gateway
can interfere with QSIG content and cause unexpected behavior with Call Back
capabilities. To prevent interference with the QSIG content, turn off the RPID
headers on the SIP gateway.
|
Default
value: None
|
QSIG Variant
|
To display the options in the QSIG Variant drop-down list box,
select QSIG from the Tunneled Protocol pulldown menu. This parameter specifies
the protocol profile that is sent in outbound QSIG facility information
elements.
From the
pulldown menu, select one of
Default
value: No Changes
|
ASN.1 ROSE OID Encoding
|
To display the options in the ASN.1 ROSE OID Encoding pulldown
menu, choose QSIG from the Tunneled Protocol pulldown menu. This parameter
specifies how to encode the Invoke Object ID (OID) for remote operations
service element (ROSE) operations.
From the
pulldown menu, select one of
- No Changes—Keep this
parameter set to the default value unless a Cisco support engineer instructs
otherwise.
-
Not
Selected
-
Use
Global Value ECMA—If you selected the ECMA option from the QSIG Variant
pulldown menu, select this option.
-
Use
Global Value ISO—If you selected the ISO option from the QSIG Variant pulldown
menu, select this option.
-
Use
Local Value
Default
value: No Changes
|
Packet Capture Mode
|
This
setting exists for troubleshooting encryption only; packet capturing may cause
high CPU usage or call-processing interruptions.
From the
pulldown menu, select one of
-
None—This option, which serves as the default setting, indicates
that no packet capturing is occurring. After you complete packet capturing,
configure this setting.
-
Batch Processing Mode—Cisco Unified
Communications Manager
writes the decrypted or nonencrypted messages to a file, and the system
encrypts each file. On a daily basis, the system creates a new file with a new
encryption key.
Cisco Unified
Communications Manager,
which stores the file for seven days, also stores the keys that encrypt the
file in a secure location.
Cisco Unified
Communications Manager
stores the file in the PktCap virtual directory. A single file contains the
time stamp, source IP address, source IP port, destination IP address, packet
protocol, message length, and the message. The TAC debugging tool uses HTTPS,
administrator username and password, and the specified day to request a single
encrypted file that contains the captured packets. Likewise, the tool requests
the key information to decrypt the encrypted file. Before you contact TAC, you
must capture the SRTP packets by using a sniffer trace between the affected
devices.
Default
value: None
|
Packet Capture Duration
(Optional)
|
This setting exists for troubleshooting encryption only; packet
capturing may cause high CPU usage or call-processing interruptions. This field
specifies the maximum number of minutes that is allotted for one session of
packet capturing.
To
initiate packet capturing, enter a value other than 0 in the field. After
packet capturing completes, the value, 0, displays.
Default
value: 0 (zero), Range is from 0 to 300 minutes
|
Media Termination Point
Required
(Optional)
|
You can
configure
Cisco Unified
Communications Manager
SIP trunks to always use an Media Termination Point (MTP). Check this box to
provide media channel information in the outgoing INVITE request. When this
check box is checked, all media channels must terminate and reoriginate on the
MTP device. If you uncheck the check box, the
Cisco Unified
Communications Manager
can decide whether calls are to go through the MTP device or be connected
directly between the endpoints.
Note
|
If the
check box remains unchecked,
Cisco Unified
Communications Manager
attempts to dynamically allocate an MTP if the DTMF methods for the call legs
are not compatible. For example, existing phones that run SCCP support only
out-of-band DTMF, and existing phones that run SIP support RFC2833. Because the
DTMF methods are not identical, the
Cisco Unified
Communications Manager
dynamically allocates an MTP. If, however, a new phone that runs SCCP, which
supports RFC2833 and out-of band, calls an existing phone that runs SIP,
Cisco Unified
Communications Manager
does not allocate an MTP because both phones support RFC2833. So, by having the
same type of DTMF method supported on each phone, there is no need for MTP.
|
Default
value: False (Unchecked)
|
Retry Video Call as Audio
(Optional)
|
This
check box pertains to outgoing SIP trunk calls and does not impact incoming
calls. By default, the system checks this check box to specify that this device
should immediately retry a video call as an audio call (if it cannot connect as
a video call) prior to sending the call to call control for rerouting. If you
uncheck this check box, a video call that fails to connect as video does not
try to establish as an audio call. The call then fails to call control, and
call control routes the call using Automatic Alternate Routing (AAR) and route
list or hunt list.
Default
value: True (Checked)
|
Path Replacement Support
(Optional)
|
This
check box is relevant when you select QSIG from the Tunneled Protocol pulldown
menu. This setting works with QSIG tunneling to ensure that non-SIP information
gets sent on the leg of the call that uses path replacement.
Default
value: False (Unchecked)
|
Transmit UTF-8 for
Calling Party Name
(Optional)
|
This
device uses the user locale setting of the device pool to determine whether to
send unicode and whether to translate received Unicode information. For the
sending device, if you check this check box and the user locale setting in the
device pool matches the terminating phone user locale, the device sends
unicode. If the user locale settings do not match, the device sends ASCII. The
receiving device translates incoming unicode characters based on the user
locale setting of the sending device pool. If the user locale setting matches
the terminating phone user locale, the phone displays the characters.
Note
|
The
phone may display malformed characters if the two ends of the trunk are
configured with user locales that do not belong to the same language group.
|
Default
value: False (Unchecked)
|
Transmit UTF-8 Names for
QSIG APDU
(Optional)
|
This device uses the user locale setting of the device pool to
determine whether to send unicode and whether to translate received Unicode
information. For the sending device, if you check this check box and the user
locale setting in the device pool matches the terminating phone user locale,
the device sends unicode and encodes in UTF-8 format. If the user locale
settings do not match, the device sends ASCII and encodes in UTF-8 format. If
the configuration parameter is not set and the user locale setting in the
device pool matches the terminating phone user locale, the device sends unicode
(if the name uses 8 bit format) and encodes in ISO8859-1 format.
Default
value: False (Unchecked)
|
Unattended Port
(Optional)
|
Check
this check box if calls can be redirected and transferred to an unattended
port, such as a voice mail port.
Default
value: False (Unchecked)
|
SRTP Allowed
(Optional)
|
Check
this check box if you want
Cisco Unified
Communications Manager
to allow secure and nonsecure media calls over the trunk. Checking this check
box enables Secure Real-Time Protocol (SRTP) SIP Trunk connections and also
allows the SIP trunk to fall back to Real-Time Protocol (RTP) if the endpoints
do not support SRTP. If you do not check this check box,
Cisco Unified
Communications Manager
prevents SRTP negotiation with the trunk and uses RTP negotiation instead.
Caution
|
If you
check this check box, Cisco strongly recommends that you use an encrypted TLS
profile, so that keys and other security related information do not get exposed
during call negotiations. If you use a non-secure profile, SRTP still works but
the keys get exposed in signaling and traces. In that case, you must ensure the
security of the network between
Cisco Unified
Communications Manager
and the destination side of the trunk.
|
Default
value: False (Unchecked)
|
Consider Traffic on This
Trunk Secure
|
This
field provides an extension to the existing security configuration on the SIP
trunk, which enables a SIP trunk call leg to be considered secure if SRTP is
negotiated, independent of the signaling transport.
From the
pulldown menu, select one of
Default
value: When using both sRTP and TLS
|
Route Class Signaling
Enabled
|
From the
pulldown menu, enable or disable route class signaling for the port. Route
class signaling communicates special routing or termination requirements to
receiving devices. It must be enabled for the port to support the Hotline
feature.
From the
pulldown menu, select one of
-
Default—The device uses the setting from the Route Class
Signaling service parameter
-
Off—Enables route class signaling. This setting overrides the
Route Class Signaling service parameter
-
On—Disables route class signaling. This setting overrides the
Route Class Signaling service parameter.
Default
value: Default
|
Use Trusted Relay Point
(Mandatory)
|
From
the pulldown menu, enable or disable whether
Cisco Unified
Communications Manager
inserts a trusted relay point (TRP) device with this media endpoint. A Trusted
Relay Point (TRP) device designates an MTP or transcoder device that is labeled
as Trusted Relay Point.
Cisco Unified
Communications Manager
places the TRP closest to the associated endpoint device if more than one
resource is needed for the endpoint (for example, a transcoder or RSVPAgent).
If both TRP and MTP are required for the endpoint, TRP gets used as the
required MTP. If both TRP and RSVPAgent are needed for the endpoint,
Cisco Unified
Communications Manager
first tries to find an RSVPAgent that can also be used as a TRP. If both TRP
and transcoder are needed for the endpoint,
Cisco Unified
Communications Manager
first tries to find a transcoder that is also designated as a TRP.
Select
one of
-
Default—The device uses the Use Trusted Relay Point setting from
the common device configuration with which this device associates
-
Off—Disables the use of a TRP with this device. This setting
overrides the Use Trusted Relay Point setting in the common device
configuration with which this device associates.
-
On—Enables the use of a TRP with this device. This setting
overrides the Use Trusted Relay Point setting in the common device
configuration with which this device associates.
Default
value: Default
|
PSTN Access
(Optional)
|
If you
use the Cisco Intercompany Media Engine feature, check this check box to
indicate that calls made through this trunk might reach the PSTN. Check this
check box even if all calls through this trunk device do not reach the PSTN.
For example, check this check box for tandem trunks or an H.323 gatekeeper
routed trunk if calls might go to the PSTN. When checked, this check box causes
the system to create upload voice call records (VCRs) to validate calls made
through this trunk device.
Default
value: True (Checked)
|
Run On All Active Unified
CM Nodes
(Optional)
|
Check
this check box to enable the trunk to run on every node.
Default
value: False (Unchecked)
|
|
Step 8
| From the
Call Routing
General tab, modify the following fields as required.
Option |
Description |
Remote-Party-ID
(Optional)
|
Use this
check box to allow or disallow the SIP trunk to send the Remote-Party-ID (RPID)
header in outgoing SIP messages from
Cisco Unified
Communications Manager
to the remote destination. If you check this box, the SIP trunk always sends
the RPID header. If you do not check this box, the SIP trunk does not send the
RPID header.
Note
|
Be
aware that Calling Name Presentation, Connected Line ID, and Connected Name
Presentation are not available when QSIG tunneling is enabled.
|
Outgoing SIP Trunk
Calls
The
configured values of the Calling Line ID Presentation and Calling Name
Presentation provide the basis for the construction of the Privacy field of the
RPID header. Each of these two options can have the values of Default, Allowed,
or Restricted. If either option is set to Default, the corresponding
information (Calling Line ID Presentation and/or Calling Name Presentation) in
the RPID header comes from the Call Control layer (which is based on
call-by-call configuration) within
Cisco Unified
Communications Manager.
If either option is set to Allowed or Restricted, the corresponding information
in the RPID header comes from the SIP trunk configuration window.
Incoming SIP Trunk
Calls
The
configured values of the Connected Line ID Presentation and Connected Name
Presentation provide the basis for the construction of the Privacy field of the
RPID header. Each of these two options can have the values of Default, Allowed,
or Restricted.
Be aware
that the Connected Line ID Presentation and Connected Name Presentation options
are relevant for 180/200 messages that the SIP trunk sends in response to
INVITE messages that
Cisco Unified
Communications Manager
receives. If either option is set to Default, the corresponding information
(Connected Line ID Presentation and/or Connected Name Presentation) in the RPID
header comes from the Call Control layer (which is based on call-by-call
configuration) within
Cisco Unified
Communications Manager.
If either option is set to Allowed or Restricted, the corresponding information
in the RPID header comes from the SIP trunk configuration window.
Note
|
The
Remote-party ID and Asserted Identity options represent independent mechanisms
for communication of display-identity information.
|
Default
value: True (Checked)
|
Asserted-Identity
(Optional)
|
Use
this check box to allow or disallow the SIP trunk to send the Asserted-Type and
SIP Privacy headers in SIP messages. If you check this check box, the SIP trunk
always sends the Asserted-Type header; whether or not the SIP trunk sends the
SIP Privacy header depends on the SIP Privacy configuration.
Outgoing SIP Trunk
Calls—P Headers
The
decision of which Asserted Identity (either P-Asserted Identity or
P-Preferred-Identity) header gets sent depends on the configured value of the
Asserted-Type option. A non-default value for Asserted-Type overrides values
that come from
Cisco Unified
Communications Manager
Call Control. If the Asserted-Type option is set to Default, the value of
Screening Identification that the SIP trunk receives from
Cisco Unified
Communications Manager
Call Control dictates the type of Asserted-Identity.
Outgoing SIP Trunk
Calls—SIP Privacy Header
The SIP
Privacy header gets used only when you check the Asserted Identity check box
and when the SIP trunk sends either a Privacy-Asserted Identity (PAI) or
Privacy Preferred Identity (PPI) header. (Otherwise the SIP Privacy header
neither gets sent nor processed in incoming SIP messages). The value of the SIP
Privacy headers depends on the configured value of the SIP Privacy option. A
non-default value for SIP Privacy overrides values that come from
Cisco Unified
Communications Manager
Call Control.
If the
SIP Privacy option is set to Default, the Calling Line ID Presentation and
Calling Name Presentation that the SIP trunk receives from
Cisco Unified
Communications Manager
Call Control determines the SIP Privacy header.
Incoming SIP Trunk
Calls—P Headers
The
decision of which Asserted Identity (either P-Asserted Identity or
P-Preferred-Identity) header gets sent depends on the configured value of the
Asserted-Type option. A non-default value for Asserted-Type overrides values
that come from
Cisco Unified
Communications Manager
Call Control. If the Asserted-Type option is set to Default, the value of
Screening Identification that the SIP trunk receives from
Cisco Unified
Communications Manager
Call Control dictates the type of Asserted-Identity.
Incoming SIP Trunk
Calls—SIP Privacy Header
The SIP
Privacy header gets used only when you check the Asserted Identity check box
and when the SIP trunk sends either a PAI or PPI header. (Otherwise the SIP
Privacy header neither gets sent nor processed in incoming SIP messages.) The
value of the SIP Privacy headers depends on the configured value of the SIP
Privacy option. A non-default value for SIP Privacy overrides values that come
from
Cisco Unified
Communications Manager
Call Control.
If the
SIP Privacy option is set to Default, the Connected Line ID Presentation and
Connected Name Presentation that the SIP trunk receives from
Cisco Unified
Communications Manager
Call Control determine the SIP Privacy header.
Note
|
The
Remote-party ID and Asserted Identity options represent independent mechanisms
for communication of display-identity information.
|
Default
value: True (Checked)
|
Asserted-Type
|
From
the pulldown menu, select one of the following values to specify the type of
Asserted Identity header that SIP trunk messages should include:
-
Default—Screening information that the SIP trunk receives from
Cisco Unified
Communications Manager
Call Control determines the type of header that the SIP trunk sends.
-
PAI—The Privacy-Asserted Identity header gets sent in outgoing
SIP trunk messages; this value overrides the Screening indication value that
comes from
Cisco Unified
Communications Manager.
-
PPI—The Privacy Preferred Identity header gets sent in outoing
SIP trunk messages; this value overrides the Screening indication value that
comes from
Cisco Unified
Communications Manager.
Note
|
These
headers get sent only if the Asserted Identity check box is checked.
|
Default
value: Default
|
SIP Privacy
(Mandatory)
|
From the
pulldown menu, select one of the following values to specify the type of SIP
privacy header for SIP trunk messages to include:
-
Default—This option represents the default value; Name/Number
Presentation values that the SIP trunk receives from the
Cisco Unified
Communications Manager
Call Control compose the SIP Privacy header. For example, if Name/Number
presentation specifies Restricted, the SIP trunk sends the SIP Privacy header;
however, if Name/Number presentation specifies Allowed, the SIP trunk does not
send the Privacy header.
-
None—The SIP trunk includes the Privacy:none header and implies
Presentation allowed; this value overrides the Presentation information that
comes from
Cisco Unified
Communications Manager.
-
ID—The SIP trunk includes the Privacy:id header and implies
Presentation restricted for both name and number; this value overrides the
Presentation information that comes from
Cisco Unified
Communications Manager.
-
ID
Critical—The SIP trunk includes the Privacy:id;critical header and implies
Presentation restricted for both name and number. The label critical implies
that privacy services that are requested for this message are critical, and, if
the network cannot provide these privacy services, this request should get
rejected. This value overrides the Presentation information that comes from
Cisco Unified
Communications Manager.
Note
|
These
headers get sent only if the Asserted Identity check box is checked.
|
Default
value: Default
|
|
Step 9
| From the
Call Routing
Inbound tab, modify the following fields as required.
Option |
Description |
Significant Digits
(Mandatory)
|
Significant digits represent the number of final digits that are retained on
inbound calls. Use for the processing of incoming calls and to indicate the
number of digits that are used to route calls that are coming in to the SIP
device.
Choose
the number of significant digits to collect, from 0 to 32, or choose 99 to
indicate all digits.
Note
|
Cisco Unified
Communications Manager
counts significant digits from the right (last digit) of the number that is
called.
|
Default
value: 99
|
Connected Line ID
Presentation
(Mandatory)
|
Cisco Unified
Communications Manager
uses connected line ID presentation (COLP) as a supplementary service to
provide the calling party with the connected party number. The SIP trunk level
configuration takes precedence over the call-by-call configuration.
Select
one of
-
Default—Allowed. Choose Default if you want
Cisco Unified
Communications Manager
to send connected line information. If a call that originates from an IP phone
on
Cisco Unified
Communications Manager
encounters a device, such as a trunk, gateway, or route pattern, that has the
Connected Line ID Presentation set to Default, the presentation value is
automatically set to Allowed.
-
Restricted—Choose Restricted if you do not want
Cisco Unified
Communications Manager
to send connected line information.
Note
|
Be
aware that this service is not available when QSIG tunneling is enabled.
|
Default
value: Default
|
Connected Name
Presentation
(Mandatory)
|
Cisco Unified
Communications Manager
uses connected name ID presentation (CONP) as a supplementary service to
provide the calling party with the connected party name. The SIP trunk level
configuration takes precedence over the call-by-call configuration.
Select
one of
-
Default—Allowed. Choose Default if you want
Cisco Unified
Communications Manager
to send connected name information.
-
Restricted—Choose Restricted if you do not want
Cisco Unified
Communications Manager
to send connected name information.
Note
|
Be
aware that this service is not available when QSIG tunneling is enabled.
|
Default
value: Default
|
Calling Search Space
(Optional)
|
From
the pulldown menu, choose the appropriate calling search space for the trunk.
The calling search space specifies the collection of route partitions that are
searched to determine how to route a collected (originating) number.
You can
configure the number of items that display in this pulldown menu by using the
Max List Box Items enterprise parameter. If more calling search spaces exist
than the Max List Box Items enterprise parameter specifies, the Find button
displays next to the drop-down list box. Click the Find button to display the
Find and List Calling Search Spaces window. Find and choose a calling search
space name.
Note
|
To set
the maximum list box items, choose
System > Enterprise Parameters and choose
CCMAdmin Parameters.
|
Default
value: None
|
AAR Calling Search Space
(Optional)
|
Choose
the appropriate calling search space for the device to use when performing
automated alternate routing (AAR). The AAR calling search space specifies the
collection of route partitions that are searched to determine how to route a
collected (originating) number that is otherwise blocked due to insufficient
bandwidth.
Default
value: None
|
Prefix DN
(Optional)
|
Enter
the prefix digits that are appended to the called party number on incoming
calls.
Cisco Unified
Communications Manager
adds prefix digits after first truncating the number in accordance with the
Significant Digits setting. You can enter the international escape character +.
Default
value: None
|
Redirecting Diversion
Header - Delivery Inbound
(Optional)
|
Check
this check box to accept the Redirecting Number in the incoming INVITE message
to the
Cisco Unified
Communications Manager.
Uncheck
the check box to exclude the Redirecting Number in the incoming INVITE message
to the
Cisco Unified
Communications Manager.
You use
Redirecting Number for voice messaging integration only. If your configured
voice-messaging system supports Redirecting Number, you should check the check
box.
Default
value: False (Unchecked)
|
Incoming Calling Party -
Prefix
(Optional)
|
Cisco Unified
Communications Manager
applies the prefix that you enter in this field to calling party numbers that
use Unknown for the Calling Party Numbering Type. You can enter up to 8
characters, which include digits, the international escape field, you cannot
configure the Strip Digits field. In this case,
Cisco Unified
Communications Manager
takes the configuration for the Prefix and Strip Digits fields from the device
pool that is applied to the device. If the word, Default, displays in the
Prefix field in the Device Pool Configuration window,
Cisco Unified
Communications Manager
applies the service parameter configuration for the incoming calling party
prefix, which supports both the prefix and strip digit functionality.
Default
value: None
|
Incoming Calling Party -
Strip Digits
(Optional)
|
Enter
the number of digits, up to the number 24, that you want
Cisco Unified
Communications Manager
to strip from the calling party number of Unknown type before it applies the
prefixes.
Default
value: None
|
Incoming Calling Party -
Calling Search Space
(Optional)
|
This
setting allows you to globalize the calling party number of Unknown calling
party number type on the device. Make sure that the calling party
transformation CSS that you choose contains the calling party transformation
pattern that you want to assign to this device. Before the call occurs, the
device must apply the transformation by using digit analysis. If you configure
the CSS as None, the transformation does not match and does not get applied.
Ensure that you configure the calling party transformation pattern in a
non-null partition that is not used for routing.
Default
value: None
|
Incoming Calling Party -
Use Device Pool CSS
(Optional)
|
Check
this check box to use the calling search space for the Unknown Number field
that is configured in the device pool that is applied to the device.
Default
value: True (Checked)
|
Incoming Called Party -
Prefix
(Optional)
|
Cisco Unified
Communications Manager
applies the prefix that you enter in this field to called numbers that use
Unknown for the Called Party Number Type. You can enter up to 16 characters,
which include digits, the international escape character (+), asterisk (*), or
the pound sign (#). You can enter the word, Default, instead of entering a
prefix.
Tip
|
If the
word, Default, displays in the Prefix field, you cannot configure the Strip
Digits field. In this case,
Cisco Unified
Communications Manager
takes the configuration for the Prefix and Strip Digits fields from the device
pool that is applied to the device. If the word, Default, displays in the
Prefix field in the Device Pool Configuration window,
Cisco Unified
Communications Manager
does not apply any prefix or strip digit functionality.
|
Default
value: None
|
Incoming Called Party -
Strip Digits
(Optional)
|
Enter
the number of digits that you want
Cisco Unified
Communications Manager
to strip from the called party number of Unknown type before it applies the
prefixes.
Tip
|
To
configure the Strip Digits field, you must leave the Prefix field blank or
enter a valid configuration in the Prefix field. To configure the Strip Digits
fields in these windows, do not enter the word, Default, in the Prefix field.
|
Default
value: None
|
Incoming Called Party -
Calling Search Space
(Optional)
|
This
setting allows you to transform the called party number of Unknown called party
number type on the device. If you choose None, no transformation occurs for the
incoming called party number. Make sure that the calling search space that you
choose contains the called party transformation pattern that you want to assign
to this device.
Default
value: None
|
Incoming Called Party -
Use Device Pool CSS
(Optional)
|
Check
this check box to use the calling search space for the Unknown Number field
that is configured in the device pool that is applied to the device.
Default
value: True (Checked)
|
Connected Party
Transformation CSS
(Optional)
|
This
setting is applicable only for inbound calls. This setting allows you to
transform the connected party number on the device to display the connected
number in another format, such as a DID or E164 number.
Cisco Unified
Communications Manager
includes the transformed number in the headers of various SIP messages,
including 200 OK and mid-call update and reinvite messages. Make sure that the
Connected Party Transformation CSS that you choose contains the connected party
transformation pattern that you want to assign to this device.
Note
|
If you
configure the Connected Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the Calling
Party Transformation pattern used for Connected Party Transformation in a
non-null partition that is not used for routing.
|
Default
value: None
|
Use Device Pool Connected
Party Transformation CSS
(Optional)
|
To use
the Connected Party Transformation CSS that is configured in the device pool
that is assigned to this device, check this check box. If you do not check this
check box, the device uses the Connected Party Transformation CSS that you
configured for this device in the Trunk Configuration window.
Default
value: True (Checked)
|
|
Step 10
| From the
Call Routing
Outbound tab, modify the following fields as required.
Option |
Description |
Called Party Transformation
CSS
(Optional)
|
This
setting allows you to send the transformed called party number in an INVITE
message for outgoing calls made over SIP Trunk. Make sure that the Called Party
Transformation CSS that you choose contains the called party transformation
pattern that you want to assign to this device.
Note
|
If you
configure the Called Party Transformation CSS as None, the transformation does
not match and does not get applied. Ensure that you configure the Called Party
Transformation CSS in a non-null partition that is not used for routing.
|
Default
value: None
|
Use Device Pool Called
Party Transformation CSS
(Optional)
|
To use
the Called Party Transformation CSS that is configured in the device pool that
is assigned to this device, check this check box. If you do not check this
check box, the device uses the Called Party Transformation CSS that you
configured for this device in the Trunk Configuration window.
Default
value: True (Checked)
|
Calling Party
Transformation CSS
(Optional)
|
This
setting allows you to send the transformed calling party number in an INVITE
message for outgoing calls made over a SIP Trunk. Also when redirection occurs
for outbound calls, this CSS is used to transform the connected number that is
sent from
Cisco Unified
Communications Manager
side in outgoing reINVITE / UPDATE messages. Make sure that the Calling Party
Transformation CSS that you choose contains the calling party transformation
pattern that you want to assign to this device.
Tip
|
If you
configure the Calling Party Transformation CSS as None, the transformation does
not match and does not get applied. Ensure that you configure the Calling Party
Transformation Pattern in a non-null partition that is not used for routing.
|
Default
value: None
|
Use Device Pool Calling
Party Transformation CSS
(Optional)
|
To use
the Calling Party Transformation CSS that is configured in the device pool that
is assigned to this device, check this check box. If you do not check this
check box, the device uses the Calling Party Transformation CSS that you
configured in the Trunk Configuration window.
Default
value: True (Checked)
|
Calling Party Selection
(Mandatory)
|
Choose
the directory number that is sent on an outbound call. Select one of the
following options to specify which directory number is sent:
-
Originator—Send the directory number of the calling device
-
First Redirect Number—Send the directory number of the
redirecting device.
-
Last
Redirect Number—Send the directory number of the last device to redirect the
call.
-
First Redirect Number (External)—Send the external directory
number of the redirecting device
-
Last
Redirect Number (External)—Send the external directory number of the last
device to redirect the call.
Default
value: Originator
|
Calling Line ID
Presentation
(Mandatory)
|
Cisco Unified
Communications Manager
uses calling line ID presentation (CLIP) as a supplementary service to provide
the calling party number. The SIP trunk level configuration takes precedence
over the call-by-call configuration.
Select
one of
-
Default—Allowed. Choose Default if you want
Cisco Unified
Communications Manager
to send calling number information.
-
Restricted—Choose Restricted if you do not want
Cisco Unified
Communications Manager
to send the calling number information.
Default
value: Default
|
Calling Name Presentation
(Mandatory)
|
Cisco Unified
Communications Manager
used calling name ID presentation (CNIP) as a supplementary service to provide
the calling party name. The SIP trunk level configuration takes precedence over
the call-by-call configuration.
Select
one of
-
Default—Allowed. Choose Default if you want
Cisco Unified
Communications Manager
to send calling name information.
-
Restricted—Choose Restricted if you do not want
Cisco Unified
Communications Manager
to send the calling name information.
Note
|
This
service is not available when QSIG tunneling is enabled.
|
Default
value: Default
|
Calling and Connected
Party Info Format
(Mandatory)
|
This
option allows you to configure whether
Cisco Unified
Communications Manager
inserts a directory number, a directory URI, or a blended address that includes
both the directory number and directory URI in the SIP identity headers for
outgoing SIP messages.
From the
pulldown menu, select one of:
-
Deliver DN only in connected party—In outgoing SIP messages,
Cisco Unified
Communications Manager
inserts the calling party’s directory number in the SIP contact header
information.
-
Deliver URI only in connected party, if available—In outgoing
SIP messages,
Cisco Unified
Communications Manager
inserts the sending party’s directory URI in the SIP contact header. If a
directory URI is not available,
Cisco Unified
Communications Manager
inserts the directory number instead.
-
Deliver URI and DN in connected party, if available—In outgoing
SIP messages,
Cisco Unified
Communications Manager
inserts a blended address that includes the calling party's directory URI and
directory number in the SIP contact headers. If a directory URI is not
available,
Cisco Unified
Communications Manager
includes the directory number only.
Note
|
You
should set this field to Deliver URI only in connected party or Deliver URI and
DN in connected party only if you are setting up URI dialing between
Cisco Unified
Communications Manager
systems of Release 9.0 or greater, or between a Cisco Unified Communications
Manager system of Release 9. 0 or greater and a third party solution that
supports URI dialing. Otherwise, you must set this field to Deliver DN only in
connected party.
|
Default
value: Deliver DN only in connected party
|
Redirecting Diversion
Header Delivery - Outbound
(Optional)
|
Check
this check box to include the Redirecting Number in the outgoing INVITE message
from the
Cisco Unified
Communications Manager
to indicate the original called party number and the redirecting reason of the
call when the call is forwarded.
Uncheck
the check box to exclude the first Redirecting Number and the redirecting
reason from the outgoing INVITE message. Use Redirecting Number for
voice-messaging integration only. If your configured voice messaging system
supports Redirecting Number, check the check box.
Default
value: False (Unchecked)
|
Caller Information Caller
ID DN
(Optional)
|
Enter
the pattern, from 0 to 24 digits that you want to use to format the Called ID
on outbound calls from the trunk. For example, in North America:
-
555XXXX = Variable Caller ID, where X represents an extension
number. The Central Office (CO) appends the number with the area code if you do
not specify it.
-
5555000 = Fixed Caller ID. Use this form when you want the
Corporate number to be sent instead of the exact extension from which the call
is placed. The CO appends the number with the area code if you do not specify
it.
You can
also enter the international escape character +.
Default
value: None
|
Caller Information -
Caller Name
(Optional)
|
Enter a
caller name to override the caller name that is received from the originating
SIP Device.
Default
value: None
|
Caller Information -
Maintain Original Caller ID DN and Caller Name in Identity Headers
(Optional)
|
This
check box is used to specify whether you will use the caller ID and caller name
in the URI outgoing request. If you check this check box, the caller ID and
caller name is used in the URI outgoing request. If you do not check this check
box, the caller ID and caller name is not used in the URI outgoing request.
Default
value: False (Unchecked)
|
|
Step 11
| From the
SP Info
tab, modify the following fields to as required.
Option |
Description |
Destination Address is an
SRV
(Optional)
|
This
field specifies that the configured Destination Address is an SRV record.
Default
value: False (Unchecked)
|
Destination - Destination
Address IPv4
(Mandatory)
|
The
Destination Address IPv4 represents the remote SIP peer with which this trunk
will communicate. The allowed values for this field are an IP address, a fully
qualified domain name (FQDN), or DNS SRV record only if the Destination Address
is an SRV field is checked.
Tip
|
For
SIP trunks that can support IPv6 or IPv6 and IPv4 (dual stack mode), configure
the Destination Address IPv6 field in addition to the Destination Address
field.
|
Note
|
SIP
trunks only accept incoming requests from the configured Destination Address
and the specified incoming port that is specified in the SIP Trunk Security
Profile that is associated with this trunk.
|
Note
|
For
configuring SIP trunks when you have multiple device pools in a cluster, you
must configure a destination address that is a DNS SRV destination port. Enter
the name of a DNS SRV port for the Destination Address and check the
Destination Address is an SRV Destination Port check box.
|
If the
remote end is a
Cisco Unified
Communications Manager
cluster, DNS SRV represents the recommended choice for this field. The DNS SRV
record should include all
Cisco Unified
Communications Managers
within the cluster.
Default
value: None
|
Destination - Destination
Address IPv6
(Mandatory if Destination - Destination Address IPv4 field above
is not completed)
|
The
Destination IPv6 Address represents the remote SIP peer with which this trunk
will communicate. You can enter one of the following values in this field:
-
A
fully qualified domain name (FQDN)
-
A
DNS SRV record, but only if the Destination Address is an SRV field is checked.
SIP
trunks only accept incoming requests from the configured Destination IPv6
Address and the specified incoming port that is specified in the SIP Trunk
Security Profile that is associated with this trunk.
If the
remote end is a
Cisco Unified
Communications Manager
cluster, consider entering the DNS SRV record in this field. The DNS SRV record
should include all
Cisco Unified
Communications Managers
within the cluster.
Tip
|
For
SIP trunks that run in dual-stack mode or that support an IP Addressing Mode of
IPv6 Only, configure this field. If the SIP trunk runs in dual-stack mode, you
must also configure the Destination Address field.
|
Default
value: None. If IPv4 field above is completed, this field can be left blank.
|
Destination - Destination
port
(Mandatory)
|
Choose
the destination port. Ensure that the value that you enter specifies any port
from 1024 to 65535, or 0.
Note
|
You
can now have the same port number that is specified for multiple trunks.
|
You do
not need to enter a value if the destination address is a DNS SRV port. The
default 5060 indicates the SIP port.
Default
value: 5060
|
Sort Order
(Mandatory)
|
Indicate
the order in which the prioritize multiple destinations. A lower sort order
indicates higher priority. This field requires an integer value.
Default
value: Empty
|
MTP Preferred Originating
Codec
(Mandatory)
|
Indicate the preferred outgoing codec by selecting one of:
-
711ulaw
-
711alaw
-
G729/G729a
-
G729b/G729ab
Note
|
To
configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or
transcoder that supports the G.729 codec.
|
This
field is used only when the MTP Termination Point Required check box is
checked.
Default
value: 711ulaw
|
BLF Presence Group
(Mandatory)
|
Configure this field with the Presence feature. From the pulldown menu, select
a Presence group for the SIP trunk. The selected group specifies the
destinations that the device/application/server that is connected to the SIP
trunk can monitor.
-
Standard Presence group is configured with installation.
Presence groups that are configured in
Cisco Unified
Communications Manager
Administration also appear in the pulldown menu.
-
Presence authorization works with presence groups to allow or
block presence requests between groups.
Tip
|
You
can apply a presence group to the SIP trunk or to the application that is
connected to the SIP trunk. If a presence group is configured for both a SIP
trunk and SIP trunk application, the presence group that is applied to the
application overrides the presence group that is applied to the trunk.
|
Default
value: Standard Presence Group
|
SIP Trunk Security
Profile
(Mandatory)
|
Select
the security profile to apply to the SIP trunk.
You must
apply a security profile to all SIP trunks that are configured in
Cisco Unified
Communications Manager
Administration. Installing Cisco Unified Communications Manager provides a
predefined, nonsecure SIP trunk security profile for autoregistration. To
enable security features for a SIP trunk, configure a new security profile and
apply it to the SIP trunk. If the trunk does not support security, choose a
nonsecure profile.
Default
value: Non Secure SIP Trunk Profile
|
Rerouting Calling Search
Space
(Optional)
|
Calling
search spaces determine the partitions that calling devices can search when
they attempt to complete a call. The rerouting calling search space gets used
to determine where a SIP user (A) can refer another user (B) to a third party
(C). After the refer is completed, B and C connect. In this case, the rerouting
calling search space that is used is that of the initial SIP user (A).
Calling
Search Space also applies to 3xx redirection and INVITE with Replaces features.
Default
value: None
|
Out-Of-Dialog Refer
Calling Search Space
(Optional)
|
Calling
search spaces determine the partitions that calling devices can search when
they attempt to complete a call. The out-of-dialog calling search space gets
used when a
Cisco Unified
Communications Manager
refers a call (B) that is coming into SIP user (A) to a third party (C) when no
involvement of SIP user (A) exists. In this case, the system uses the out-of
dialog calling search space of SIP user (A).
Default
value: None
|
SUBSCRIBE Calling Search
Space
(Optional)
|
Supported with the Presence feature, the SUBSCRIBE calling search space
determines how
Cisco Unified
Communications Manager
routes presence requests from the device/server/application that connects to
the SIP trunk. This setting allows you to apply a calling search space separate
from the call-processing search space for presence (SUBSCRIBE) requests for the
SIP trunk.
From the
pulldown menu, choose the SUBSCRIBE calling search space to use for presence
requests for the SIP trunk. All calling search spaces that you configure in
Cisco Unified
Communications Manager
Administration display in the SUBSCRIBE Calling Search Space pulldown menu.
If you
do not select a different calling search space for the SIP trunk from the
pulldown menu, the SUBSCRIBE calling search space defaults to None.
To
configure a SUBSCRIBE calling search space specifically for this purpose,
configure a calling search space as you do all calling search spaces.
Default
value: None
|
SIP Profile
(Mandatory)
|
From
the drop-down list box, select the SIP profile that is to be used for this SIP
trunk.
Default
value: Standard SIP Profile
|
DTMF Signaling Method
(Mandatory)
|
Select
one of
-
No
Preference—Cisco Unified
Communications Manager
picks the DTMF method to negotiate DTMF, so the call does not require an MTP.
If Cisco Unified Communications Manager has no choice but to allocate an MTP
(if the Media Termination Point Required check box is checked), SIP trunk
negotiates DTMF to RFC2833.
-
RFC
2833—Choose this configuration if the preferred DTMF method to be used across
the trunk is RFC2833.
Cisco Unified
Communications Manager
makes every effort to negotiate RFC2833, regardless of MTP usage. Out of band
(OOB) provides the fallback method if the peer endpoint supports it.
-
OOB
and RFC 2833—Choose this configuration if both out of band and RFC2833 should
be used for DTMF.
Note
|
If the
peer endpoint supports both out of band and RFC2833,
Cisco Unified
Communications Manager
negotiates both out-of-band and RFC2833 DTMF methods. As a result, two DTMF
events are sent for the same DTMF keypress (one out of band and the other,
RFC2833).
|
Default
value: No Preference
|
Normalization Script
(Optional)
|
From
the pulldown menu, choose the script that you want to apply to this trunk.
To
import another script, on
Cisco Unified
Communications Manager
go to the SIP Normalization Script Configuration window (Device > Device
Settings > SIP Normalization Script), and import a new script file.
Default
value: None
|
Normalization Script -
Enable Trace
(Optional)
|
Check
this check box to enable tracing within the script or uncheck the check box to
disable tracing. When checked, the trace.output API provided to the Lua
scripter produces SDI trace.
Note
|
Cisco
recommends that you only enable tracing while debugging a script. Tracing
impacts performance and should not be enabled under normal operating
conditions.
|
Default
value: False (Unchecked)
|
Script Parameters
(Optional)
|
Enter
parameter names and values in the format
Param1Name=Param1Value; Param2Name=Param2Value where
Param1Name is the name of the first script parameter and
Param1Value is the value of the first script parameter.
Multiple parameters can be specified by putting semicolon after each name and
value pair . Valid values include all characters except equal signs (=),
semi-colons (;); and non-printable characters, such as tabs. You can enter a
parameter name with no value.
|
Recording Information
(Optional)
|
|
|
Step 12
| From the
GeoLocation tab, modify the following fields as required.
Option |
Description |
Geolocation
(Optional)
|
From
the drop-down list box, choose a geolocation.
You can
choose the Unspecified geolocation, which designates that this device does not
associate with a geolocation.
On
Cisco Unified
Communications Manager,
you can also choose a geolocation that has been configured with the
System
> Geolocation Configuration menu option.
Default
value: None
|
Geolocation Filter
(Optional)
|
From
the pulldown menu, choose a geolocation filter.
If you
leave the <None> setting, no geolocation filter gets applied for this
device.
On
Cisco Unified
Communications Manager,
you can also choose a geolocation filter that has been configured with the
System
> Geolocation Filter
menu option.
Default
value: None
|
Send Geolocation
Information
(Optional)
|
Check
this check box to send geolocation information for this device.
Default
value: False (Unchecked)
|
|
Step 13
| Perform one
of
- To save a new SIP trunk,
click
Save.
- To save an updated SIP
trunk, click
Update.
The
SIP trunk appears in the SIP trunk list. You can view the SIP trunk and its
characteristics by logging in to the
Cisco Unified
Communications Manager
where the SIP trunk was added, selecting
Device >
Trunk, and performing the
Find
operation. When you click on the name of the SIP trunk in the list, the trunk
characteristics are displayed.
Note
|
The SIP
trunk is automatically reset on the
Cisco Unified
Communications Manager
as soon as it is added. To reset the SIP trunk at any other time, perform
Reset SIP Trunks.
|
|