This chapter provides information about Cisco Unified
Communications gateways which enable
Cisco Unified Communications Manager to communicate with non-IP telecommunications
devices.
Cisco Unified Communications Manager supports several types of voice gateways.
Gateways enable
Cisco Unified Communications Manager to communicate with non-IP telecommunications
devices. The following steps are required to configure gateways in
Cisco Unified Communications Manager.
Procedure
Step 1
Install and configure the gateway or voice gateway module in the
network.
Step 2
Gather the information that you need to configure the gateway to
operate with
Cisco Unified Communications Manager.
Step 3
On the gateway, perform any required configuration steps.
Step 4
Add and configure the gateway in
Cisco Unified Communications Manager Administration.
Step 5
Add and configure ports on the gateway or add and configure the
Cisco VG248 Analog Phone Gateway.
Step 6
For FXS ports, add directory numbers, if appropriate.
Step 7
Configure the dial plan for the gateway for routing calls out to
the PSTN or other destinations. This configuration can include setting up a
route group, route list, and route pattern for the Gateway in
Cisco Unified Communications Manager or, for some gateways, configuring the dial
plan on the gateway itself.
Step 8
Reset the gateway to apply the configuration settings.
Tip
To get to the default web pages for many gateway devices, you
can use the IP address of that gateway. Make your hyperlink url =
http://x.x.x.x/, where x.x.x.x is the dot-form IP address of the device. The
web page for each gateway contains device information and the real-time status
of the gateway.
Set up MGCP BRI gateway
The following steps are required to configure a BRI gateway
in
Cisco Unified Communications Manager.
Procedure
Step 1
Install and configure the gateway and voice modules in the
network.
Step 2
Gather the information that you need to configure the gateway to
operate with
Cisco Unified Communications Manager and to configure the trunk interface to the
PSTN or external non-IP telephony device.
Step 3
On the gateway, perform any required configuration steps.
Step 4
Add and configure the gateway in
Cisco Unified Communications Manager Administration.
Step 5
Add and configure ports on the gateway.
Step 6
Configure the dial plan for the gateway for routing calls out to
the PSTN or other destinations.
This configuration can include setting up a route group, route
list, and route pattern for the gateway in
Cisco Unified Communications Manager or, for some gateways, configuring the dial
plan on the gateway itself.
Step 7
Reset the gateway to apply the configuration settings.
Tip
To get to the default web pages for gateway devices, you can use
the IP address of that gateway. Make your hyperlink url = http://x.x.x.x/,
where x.x.x.x specifies the dot-form IP address of the device. The web page for
each gateway contains device information and the real-time status of the
gateway.
Cisco voice gateways
Cisco Unified Communications Manager supports several types of Cisco
Unified Communications gateways. Gateways use call control protocols to
communicate with the PSTN and other non-IP telecommunications devices, such as
private branch exchanges (PBXs), key systems, analog phones, fax machines, and
modems.
Trunk interfaces specify how the gateway communicates with
the PSTN or other external devices by using time-division-multiplexing (TDM)
signaling.
Cisco Unified Communications Manager and Cisco gateways use a variety of TDM
interfaces, but supported TDM interfaces vary by gateway model.
The following list provides available interfaces that
Cisco Unified Communications Manager supports with MGCP gateways:
Foreign Exchange Office (FXO)
Foreign Exchange Station (FXS)
T1 Channel Associated Signaling (CAS) recEive and transMit or ear
and mouth (E&M)
Basic Rate Interface (BRI) Q.931
T1 PRI-North American ISDN Primary Rate Interface (PRI)
E1 PRI-European ISDN Primary Rate Interface (PRI)
The following list provides available interfaces that
Cisco Unified Communications Manager supports with H.323 gateways:
FXO
FXS
E&M
Analog Direct Inward Dialing (DID)
Centralized Automatic Message Accounting (CAMA)
BRI Q.931
BRI QSIG-Q signaling protocol that is based on ISDN standards
T1 CAS FXS, FXO, and E&M
T1 FGD
T1/E1 PRI
T1 PRI NFAS
T1/E1 QSIG
E1 R2
J1
The following list provides available interfaces that
Cisco Unified Communications Manager supports with SCCP gateways:
FXS
Cisco Unified Communications Manager can use H.323 gateways that support E1
CAS, but you must configure the E1 CAS interface on the gateway.
The Cisco VG248 Analog Phone Gateway has a standalone,
19-inch rack-mounted chassis with 48-FXS ports. This product allows on-premise
analog telephones, fax machines, modems, voice-messaging systems, and
speakerphones to register with a single
Cisco Unified Communications Manager system.
Cisco VG248 Analog Phone Connectivity
The Cisco VG248 Analog Phone Gateway communicates with
Cisco Unified Communications Manager by using the Skinny Client Control Protocol to
allow support for the following supplementary services features for analog
phones:
Call transfer
Conference
Call waiting (with calling party ID display)
Hold (including switch between parties on hold)
Music on hold
Call forward all
Send all calls to voice-messaging system
Group call pickup
Voice-messaging system message waiting indication
Speed dial (maximum of 9 speed dials)
Last number redial
Cisco fax relay
Dynamic port and device status that is available from
Cisco Unified Communications Manager
Cisco VGC Phone Device Types
All Cisco VG248 ports and units appear as distinct devices
in
Cisco Unified Communications Manager with the device type
"Cisco VGC Phone."Cisco Unified Communications Manager recognizes and configures each port as a
phone.
Fax and Modem Connectivity
The Cisco VG248 supports legacy fax machines and modems.
When using fax machines, the Cisco VG248 uses either the Cisco fax relay or
pass-through/up speed technology to transfer faxes across the network with high
reliability.
You can connect any modem to the Cisco VG248 by using
pass-through mode.
Voice-Mail Connectivity
The Cisco VG248 generates call information by using the
Simplified Message Desk Interface (SMDI) format for all calls that are ringing
on any of the 48 analog lines that connect to it. It will also pass on SMDI
call information from other Cisco VG248s, or from a legacy PBX, to the
voice-messaging system. Any commands for message-waiting indicators get sent to
Cisco Unified Communications Manager and to any other attached SMDI hosts.
This mechanism allows for many new configurations when
SMDI-based voice-messaging systems are used, including
You can share a single voice-messaging system between
Cisco Unified Communications Manager and a legacy PBX.
Voice-messaging system and Cisco VG248 can function remotely in a
centralized call-processing model.
Multiple clusters can use a single voice-messaging system, by
using one Cisco VG248 per cluster.
Configure multiple voice-messaging systems in a single cluster
because the Cisco VG248 generates SMDI call information rather than the
Cisco Unified Communications Manager.
Cisco VG248 Time Device
The Cisco VG248 contains a real-time clock that is
persistent across power cycles and restarts. The real-time clock gets set for
the first time when the device registers with
Cisco Unified Communications Manager. The clock gets set by using the
DefineDateTime Skinny message that
Cisco Unified Communications Manager sends. After a power cycle or restart, the
clock resets when the Cisco VG248 receives the DefineDateTime message from
Cisco Unified Communications Manager and then resets no more than once per hour
thereafter.
Cisco VG248 Configuration File Updates
The Cisco VG248 queries the TFTP server to access the
configuration files for the device. The configuration files update whenever you
modify the configuration of the Cisco VG248 via
Cisco Unified Communications Manager.
Cisco VG224 Analog Phone Gateway
The Cisco VG224 Analog Phone Gateway, which has a standalone, 17-inch rack-mounted chassis with 24-FXS ports, allows on-premise analog telephones, fax machines, modems, and speakerphones to register with Cisco Unified Communications Manager.
This gateway supports the H.323, MGCP, SCCP, SIP, and T.38 fax relay.
Cisco Voice Gateway 200
The Cisco Unified Communications Voice Gateway (VG200)
provides a 10/100BaseT Ethernet port for connection to the data network. The
following list gives available telephony connections:
1 to 4 FXO ports for connecting to a central office or PBX
1 to 4 FXS ports for connecting to POTS telephony devices
1 or 2 Digital Access T1 ports for connecting to the PSTN
1 or 2 Digital Access PRI ports for connecting to the PSTN
MGCP or H.323 interface to
Cisco Unified Communications Manager
MGCP mode supports T1/E1 PRI, T1 CAS, FXS, FXO. (Only the user
side supports BRI.)
H.323 mode supports E1/T1 PRI, E1/T1 CAS, FXS, and FXO. H.323
mode supports E&M, fax relay, and G.711 modem.
The MGCP VG200 integration with legacy voice-messaging
systems allows the
Cisco Unified Communications Manager to associate a port with a voice mailbox and
connection.
MGCP BRI call connections
Previously, gateways used H.323 signaling to
Cisco Unified Communications Manager to provide interfaces to the public switched
telephone network (PSTN) for BRI ISDN connections.
Now,
Cisco Unified Communications Manager can use a Media Gateway Control Protocol
(MGCP) gateway to handle BRI ISDN connections to the PSTN and to provide a
centrally administered gateway interface.
Cisco Unified Communications Manager uses logical connections to exchange MGCP and
ISDN Q.931 messages with the gateway. This connection uses a User Datagram
Protocol (UDP) logical connection for exchanging MGCP messages and a
Transmission Control Protocol (TCP) connection for the backhaul ISDN Q.931
messages.
The following figure shows a typical scenario that centralizes
call processing for remote-site BRI trunk gateways that connect to the PSTN.
When a call arrives from or goes to the PSTN over the BRI trunk, the
Cisco Unified Communications Manager and the gateway (based on an IOS router)
exchange ISDN Q.931 messages across the WAN.
Figure 1. Topology Shows a Scenario by Using MGCP BRI Interfaces
Note
The BRI gateway supports MGCP BRI backhaul for BRI trunk only. It does
not support BRI phone or station. The IOS gateway supports BRI phones that use
Skinny Client Control Protocol.
Switch-based gateways
Several telephony modules for the Cisco Catalyst 4000 and 6000 family switches act as telephony gateways. You can use existing Cisco Catalyst 4000 or 6000 family devices to implement IP telephony in your network by using the following voice gateway modules:
Install Catalyst 6000 voice gateway modules that are line cards in any Cisco Catalyst 6000 or 6500 series switch.
Install the Catalyst 4000 access gateway module in any Catalyst 4000 or 4500 series switch.
Cisco Catalyst 6000 8-Port Voice T1/E1 and Services Module
The Cisco Catalyst 6000 8-Port Voice T1/E1 and Services
Modules provide the following features:
8 ports for providing
Digital T1/E1 connectivity to the PSTN (T1/E1 PRI or T1 CAS)
Digital signal processor (DSP) resources for transcoding and
conferencing
MGCP interface to
Cisco Unified Communications Manager
Connection to a voice-messaging system (using T1 CAS)
Users have the flexibility to use ports on a T1 module for
T1 connections or as network resources for voice services. Similarly, the E1
module provides ports for E1 connections or as network resources. The ports can
serve as T1/E1 interfaces, or the ports will support transcoding or
conferencing.
Note
Either module supports DSP features on any port, but T1 modules
cannot be configured for E1 ports, and E1 modules cannot be configured for T1
ports.
Similar to the Cisco MGCP-controlled gateways with FXS
ports, the Cisco 6608 T1 CAS gateway supports hookflash transfer. Hookflash
transfer defines a signaling procedure that allows a device, such as a
voice-messaging system, to transfer to another destination. While the device is
connected to
Cisco Unified Communications Manager through a T1 CAS gateway, the device performs
a hookflash procedure to transfer the call to another destination.
Cisco Unified Communications Manager responds to the hookflash by using a blind
transfer to move the call. When the call transfer completes, the voice channel
that connected the original call to the device gets released.
Note
Only E&M T1 ports support hookflash transfer.
Cisco Catalyst 6000 24 Port FXS Analog Interface Module
The Cisco Catalyst 6000 24 Port FXS Analog Interface Module provides the following features:
MGCP interface to Cisco Unified Communications Manager
The Catalyst 6000 24 Port FXS Analog Interface Module provides 24 FXS ports for connecting to analog phones, conference room speakerphones, and fax machines. You can also connect to legacy voice-messaging systems by using SMDI and by associating the ports with voice-messaging extensions.
The FXS module provides legacy analog devices with connectivity into the IP network. Analog devices can use the IP network infrastructure for toll-bypass applications and to communicate with devices such as SCCP IP phones and H.323 end stations. The FXS module also supports fax relay, which enables compressed fax transmission over the IP WAN and preserves valuable WAN bandwidth for other data applications.
Cisco Communication Media Module
The Cisco Communication Media Module (CMM), which is a Catalyst 6500 line card, provides T1 and E1 gateways that allow organizations to connect their existing TDM network to their IP communications network. The Cisco CMM provides connectivity to the PSTN also. You can configure the Cisco CMM, which provides an MGCP, H.323, or SIP interface to Cisco Unified Communications Manager, with the following interface and service modules:
6-port T1 interface module for connecting to the PSTN or a PBX
6-port E1 interface module for connecting to the PSTN or a PBX
24-port FXS interface module for connecting to POTS telephony devices
Note
The Cisco CMM fits in the Cisco 7600 platform chassis.
Cisco Catalyst 4000 Access Gateway Module
The Cisco Catalyst 4000 Access Gateway Module provides an MGCP or H.323 gateway interface to Cisco Unified Communications Manager. You can configure this module with the following interface and service modules:
6 ports for FXS and FXO
2 T1/E1 ports for Digital Access PRI and Digital Access T1
Cisco Catalyst 4224 Voice Gateway Switch
The Cisco Catalyst 4224 Voice Gateway Switch provides a single-box solution for small branch offices. The Catalyst 4224 provides switching, IP routing, and PSTN voice-gateway services by using onboard digital signal processors (DSPs). The Catalyst 4224 has four slots that you can configure with multiflex voice and WAN interface cards to provide up to 24 ports. These ports can support the following voice capabilities:
FXS ports for POTS telephony devices
FXO ports for PSTN connections
T1 or E1 ports for Digital Access PRI, and Digital Access T1 services
The Cisco Catalyst 4224 Access Gateway Switch provides an MGCP or H.323 interface to Cisco Unified Communications Manager.
H.323 Gateways
H.323 devices comply with the H.323 communications standards and enable video conferencing over LANs and other packet-switched networks. You can add third-party H.323 devices or other Cisco devices that support H.323 (such as the Cisco 2600 series, 3600 series, or 5300 series gateways).
Cisco IOS H.323 gateways such as the Cisco 2600, 3600, 1751, 1760, 3810 V3, 7200 7500, AS5300, and VG200 provide full-featured routing capabilities. See the documentation for each of these gateway types for information about supported voice gateway features and configuration.
Outbound FastStart call connections
Calls that are placed from IP phones over large WAN
topologies can experience voice clipping when the called party goes off hook to
answer the call. When H.323 trunks or gateways are separated from the
Cisco Unified Communications Manager server, significant delays can occur because
of the many H.245 messages that are exchanged when a call is set up.
With the FastStart feature, information that is required to
complete a media connection between two parties gets exchanged during the H.225
portion of call setup, and this exchange eliminates the need for H.245
messages. The connection experiences one roundtrip WAN delay during call setup,
and the calling party does not receive voice clipping when the called party
answers the call.
Cisco Unified Communications Manager uses media termination points (MTP)
for making an H.323 outbound FastStart call.
Cisco Unified Communications Manager starts an outbound FastStart call by
allocating an MTP and opening the receive channel. Next, the H.323 Fast Connect
procedure sends the SETUP message with a FastStart element to the called
endpoint. The FastStart element includes information about the receiving
channel for the MTP.
The called endpoint accepts the H.323 Fast Connect procedure
by sending a CALL PROCEEDING, PROGRESS, ALERT, or CONNECT message that contains
a FastStart element. When
Cisco Unified Communications Manager receives the FastStart element, it connects
the media immediately and avoids the delays with the usual exchange of H.245
messages.
The called endpoint can refuse the H.323 Fast Connect
procedure by not returning the FastStart element in any of the messages up to
and including the CONNECT message. In this case, the
Cisco Unified Communications Manager handles the call as a normal call and uses the
MTP for subsequent media cut-through.
The Outbound FastStart feature requires an MTP. If an MTP is
not available when the call is set up, the call continues without FastStart and
with no supplementary services. If you want all calls to use FastStart only,
you can set the service parameter called
"Fail call if MTP allocation fails," which is located in the
Cluster Wide Parameters (Device-H323) portion of the service parameters for the
Cisco Unified CallManager service. When you set this parameter to True, the
system rejects calls when no MTP is available.
Voice gateway model summary
The following table summarizes Cisco voice gateways that
Cisco Unified Communications Manager supports with information about the supported
signaling protocols, trunk interfaces, and port types.
Table 1 Overview of Supported Voice Gateways, Protocols, Trunk Interfaces,
and Port Types
Gateway Model
Supported Signaling Protocols
Trunk Interfaces
Port Types
Notes
Cisco IOS Integrated Routers
Cisco 1751 and Cisco 1760
H.323 and SIP
FXS
Loopstart or groundstart
Basic calls only
FXO
Loopstart or groundstart
E&M
Analog DID
CAMA
BRI
BRI QSIG
Basic calls only
T1 CAS (E&M, FXS, FXO)
E1 R2
T1/E1 QSIG
Basic calls only
T1/E1 PRI
MGCP
FXS
Loopstart only
Basic calls only
FXO
Loopstart or groundstart
No caller ID
BRI
User side only; no QSIG support
T1 CAS (E&M)
T1/E1 QSIG
Supplementary Services
T1/E1 PRI
Cisco 2600, 2600XM series, 2691, 3700 series
H.323 and SIP
FXS
Loopstart or groundstart
FXO
Loopstart or groundstart
Basic calls only
E&M
Analog DID
CAMA
BRI
BRI QSIG
Basic calls only
T1 CAS (E&M, FXS, FXO)
T1 FGD
E1 CAS
E1 R2
T1/E1 QSIG
Basic calls only
T1/E1 PRI
T1 PRI NFAS
T1 PRI (Megacom/SDN)
Per T1 port only; not per call
MGCP
FXS
Loopstart only
Basic calls only
FXO
Loopstart or groundstart
No caller ID
BRI
Only 2600XM/2691 support MGCP BRI; User side only; no QSIG
support
T1 CAS (E&M)
T1/E1 QSIG
Supplementary Services
T1/E1 PRI
T1 PRI (Megacom/SDN)
Per call
SCCP
BRI
DoD STE BRI phones only; single-B-channel
Cisco 3600 series
H.323 and SIP
FXS
Loopstart or groundstart
Basic calls only
FXO
Loopstart or groundstart
E&M
Analog DID
CAMA
BRI
BRI QSIG
Basic calls only
T1 CAS (E&M, FXS, FXO)
T1 FGD
E1 CAS
E1 R2
T1/E1 QSIG
Basic calls only
T1/E1 PRI
T1 PRI NFAS
T1 PRI (Megacom/SDN)
Per T1 port only; not per call
MGCP
FXS
Loopstart only
Basic calls only
FXO
Loopstart or groundstart
No caller ID
BRI
Only 3640/3660 and some interface cards support MGCP BRI; User
side only; no QSIG support
T1 CAS (E&M)
T1/E1 QSIG
Supplementary Services
T1/E1 PRI
T1 PRI (Megacom/SDN)
Per call
Cisco 2800 and 3800 series
H.323 and SIP
FXS
Loopstart or groundstart
Basic calls only
FXO
Loopstart or groundstart
E&M
Analog DID
CAMA
BRI
BRI QSIG
Basic calls only
T1 CAS (E&M, FXS, FXO)
T1 FGD
E1 CAS
E1 R2
T1/E1 QSIG
Basic calls only
T1/E1 PRI
T1 PRI NFAS
T1 PRI (Megacom/SDN)
Per T1 port only; not per call
MGCP
FXS
Loopstart only
Basic calls only
FXO
Loopstart or groundstart
No caller ID
BRI
User side only; no QSIG support
T1 CAS (EM)
T1/E1 QSIG
Supplementary Services
T1/E1 PRI
T1 PRI (Megacom/SDN)
Per call
SCCP
FXS
Cisco 7200 series
H.323
T1 CAS (E&M, FXS, FXO)
T1 FGD
E1 R2
T1/E1 QSIG
Basic calls only
T1/E1 PRI
Cisco 5000 series
H.323 and SIP
T1 CAS (E&M, FXS, FXO)
T1 FGB
T1 FGD
E1 R2
T1/E1 QSIG
Basic calls only
T1/E1 PRI
Cisco Standalone Voice Gateways
Cisco VG224 Analog Gateway
H.323, MGCP, and SIP
FXS
Basic calls only
SCCP
FXS
Supplementary Services
Cisco VG248 Analog Gateway
SCCP
FXS
Supplementary Services
Cisco VG200 Gateway
H.323 and SIP
FXS
Loopstart or groundstart
Basic calls only
FXO
Loopstart or groundstart
E&M
Analog DID
CAMA
BRI
BRI QSIG
T1 CAS (E&M, FXS, FXO)
T1 FGD
E1 CAS
E1 R2
T1/E1 QSIG
Basic calls only
T1/E1 PRI
T1 PRI NFAS
T1 PRI (Megacom/SDN)
Per T1 port only; not per call
MGCP
FXS
Loopstart only
Basic calls only
FXO
Loopstart or groundstart
No caller ID
BRI
Only 2600XM/2691 support MGCP BRI; User side only; no QSIG
support
T1 CAS (E&M)
T1/E1 QSIG
Supplementary Services
T1/E1 PRI
T1 PRI (Megacom/SDN)
Per call
Cisco Access Analog Trunk Gateway (AT-2, AT-4, AT-8)
SCCP
FXO
Loop start
Cisco Access Analog Station Gateway (AS-2, AS-4, AS-8)
Cisco Catalyst 6000 8-Port Voice T1/E1 and Services Module
(WS-X6608-T1)
MGCP
T1 CAS (E&M)
T1 PRI
T1 QSIG
Supplementary Services
Cisco Catalyst 6000 8-Port Voice T1/E1 and Services Module
(WS-X6608-E1)
MGCP
E1 PRI
E1 QSIG
Supplementary Services
Cisco Catalyst 6000 24-Port FXS Analog Interface Module
(WS-X6624-FXS)
MGCP
FXS
Loopstart only
Basic calls only
Gateways dial plans and route groups
Gateways use dial plans to access or call out to the PSTN,
route groups, and group-specific gateways. The different gateways that are used
within Cisco Unified Communications Solutions have dial plans that are
configured in different places:
Configure dial plan information for both Skinny and MGCP gateways
in the
Cisco Unified Communications Manager.
Configure dial plans in
Cisco Unified Communications Manager to access the H.323-based Cisco IOS software
gateways. Configure dial peers in the H.323-based gateways to pass the call out
of the gateway.
The route group points to one or more gateways and can
choose the gateways for call routing based on preference. The route group can
serve as a trunk group by directing all calls to the primary device and then
using the secondary devices when the primary is unavailable. One or more route
lists can point to the same route group.
All devices in a given route group share the same
characteristics such as path and digit manipulation.
Cisco Unified Communications Manager restricts the gateways that you can include in the same route
group and the route groups that you can include in the same route list.
Route groups can perform digit manipulation that will
override what was performed in the route pattern. Configuration information
that is associated with the gateway defines how the call is actually placed and
can override what was configured in the route pattern.
You can configure H.323 trunks, not H.323gateways, to be
gatekeeper-controlled trunks. This means that before a call is placed to an
H.323 device, it must successfully query the gatekeeper.
Multiple clusters for inbound and outbound calls can share
H.323 trunks, but MGCP and Skinny-based gateways remain dedicated to a single
Cisco Unified Communications Manager cluster.
Dependency records for gateways and their route groups and directory numbers
To find route groups or directory numbers that a specific
gateway or gateway port is using, click the Dependency Records link that is
provided on the
Cisco Unified Communications Manager Administration Gateway Configuration window.
The Dependency Records Summary window displays information about route groups
and directory numbers that are using the gateway or port. To find out more
information about the route group or directory number, click the route group or
directory number, and the Dependency Records Details window displays. If the
dependency records are not enabled for the system, the dependency records
summary window displays a message.
Gateways and the Local Route Groups feature
A special virtual Local Route Group can be bound to a real
route group differently based on the Local Route Group device pool setting of
the originating device. Devices, such as phones, from different locales can
therefore use identical route lists and route patterns, but
Cisco Unified Communications Manager selects the correct gateway(s) for their local
end.
If the Local Route Group feature is in use, configuration
of gateways changes, particularly with respect to configuration of the
following gateway fields:
Called Party Transformation CSS
Use Device Pool Called Party Transformation CSS
Gateways and the Calling Party Normalization feature
In line with E.164 standards, calling party normalization
enhances the dialing capabilities of some phones and improves call back
functionality when a call is routed to multiple geographical locations; that
is, the feature ensures that the called party can return a call without needing
to modify the directory number in the call log directories on the phone.
Additionally, calling party normalization allows you to globalize and localize
phone numbers, so the appropriate calling number presentation displays on the
phone.
Configuring calling party normalization alleviates issues
with toll bypass where the call is routed to multiple locations over the IP
WAN. In addition, it allows
Cisco Unified Communications Manager to distinguish the origin of the call to
globalize or localize the calling party number for the phone user.
SIP trunks and MGCP gateways can support sending the
international escape character, +, for calls. H.323 gateways/trunks do not
support the + because the H.323 protocol does not support the international
escape character, +. For outgoing calls through a gateway that supports +,
Cisco Unified Communications Manager can send the + with the dialed digits to the
gateway/trunk. For outgoing calls through a gateway/trunk that does not support
+, the international escape character + gets stripped when
Cisco Unified Communications Manager sends the call information to the
gateway/trunk.
SIP does not support the number type, so calls through SIP
trunks only support the Incoming Calling Party Unknown Number (prefix and
digits-to-strip) settings.
You can configure the international escape character, +, to
globalize the calling party number. For information on the international escape
character, +, see
Use the international escape character.
Apply the international escape character to inbound calls over H.323 trunks
The H.323 protocol does not support the international escape character, +. To ensure that correct prefixes, including the international escape character, +, get applied for inbound calls over H.323 gateways/trunks, you must configure the incoming called party settings in the service parameter, device pool, H.323 gateway, or H.323 trunk windows; that is, configuring the incoming called party settings ensures that when a inbound call comes from a H.323 gateway or trunk, Cisco Unified Communications Manager transforms the called party number back to the value that was originally sent over the trunk/gateway.
For example, to ensure that the correct DN patterns get used with SAF/call control discovery for inbound calls over H.323 gateways/trunks, you must configure the incoming called party settings in the service parameter, device pool, or H.323 (non-gatekeeper controlled) trunk window. See the following example for more information.
A caller places a call to +19721230000 to Cisco Unified Communications Manager A.
Cisco Unified Communications Manager A receives +19721230000 and transforms the number to 55519721230000 before sending the call to the H.323 trunk. In this case, your configuration indicates that the international escape character + should be stripped and 555 should be prepended for calls of International type.
For this inbound call from the trunk, Cisco Unified Communications Manager B receives 55519721230000 and transforms the number back to +19721230000 so that digit analysis can use the value as it was sent by the caller. In this case, your configuration for the incoming called party settings indicates that you want 555 to be stripped and +1 to be prepended to called party numbers of International type.
The service parameters support the Cisco CallManager service. To configure the service parameters, click Advanced in the Service Parameter Configuration window for the Cisco CallManager service; then, locate the H.323 pane for the following parameters:
Incoming Called Party National Number Prefix - H.323
Incoming Called Party International Number Prefix - H.323
Incoming Called Party Subscriber Number Prefix - H.323
Incoming Called Party Unknown Number Prefix - H.323
These service parameters allow you to prefix digits to the called number based on the Type of Number field for the inbound offered call. You can also strip a specific number of leading digits before the prefix gets applied. To prefix and strip digits by configuring these parameter fields, use the following formula, x:y, where x represents the exact prefix that you want to add to called number and y represents the number of digits stripped; be aware that the colon separates the prefix and the number of stripped digits. For example, enter 91010:6 in the field, which means that you want to strip 6 digits and then add 901010 to the beginning of the called number. In this example, a national call of 2145551234 becomes 910101234. You can strip up to 24 digits and prefix/add up to than 16 digits.
Gateway failover and fallback
This section describes how these Cisco voice gateways handle
Cisco Unified Communications Manager failover and fallback situations.
To handle Cisco Unified Communications Manager failover situations, MGCP gateways receive a list of Cisco Unified Communications Managers that is arranged according to the Cisco Unified Communications Manager group and defined for the device pool that is assigned to the gateway. A Cisco Unified Communications Manager group can contain one, two, or three Cisco Unified Communications Managers that are listed in priority order for the gateway to use. If the primary Cisco Unified Communications Manager in the list fails, the secondary Cisco Unified Communications Manager gets used. If the primary and secondary Cisco Unified Communications Managers fail, the tertiary Cisco Unified Communications Manager gets used.
Fallback describes the process of recovering a higher priority Cisco Unified Communications Manager when a gateway fails over to a secondary or tertiary Cisco Unified Communications Manager. Cisco MGCP gateways periodically take status of higher priority Cisco Unified Communications Managers. When a higher priority Cisco Unified Communications Manager is ready, it gets marked as available again. The gateway reverts to the highest available Cisco Unified Communications Manager when all calls go idle or within 24 hours, whichever occurs first. The administrator can force a fallback either by stopping the lower priority Cisco Unified Communications Manager whereby calls get preserved, by restarting the gateway, which preserves calls, or by resetting Cisco Unified Communications Manager, which terminates calls.
Note
Skinny Client Control Protocol (SCCP) gateways handle Cisco Unified Communications Manager redundancy, failover, and fallback in the same way as MGCP gateways.
IOS H.323 gateways
Cisco IOS gateways also handle Cisco Unified Communications Manager failover situations. By using several enhancements to the dial-peer and voice class commands in Cisco IOS Release 12.1(2)T, Cisco IOS gateways can support redundant Cisco Unified Communications Managers. The command, h225 tcp timeout seconds, specifies the time that it takes for the Cisco IOS gateway to establish an H.225 control connection for H.323 call setup. If the Cisco IOS gateway cannot establish an H.225 connection to the primary Cisco Unified Communications Manager, it tries a second Cisco Unified Communications Manager that is defined in another dial-peer statement. The Cisco IOS gateway shifts to the dial-peer statement with the next highest preference setting.
The following example shows the configuration for H.323 gateway failover:
interface FastEthernet0/0ip address 10.1.1.10 255.255.255.0
dial-peer voice 101 voip
destination-pattern 1111
session target ipv4:10.1.1.101
preference 0
voice class h323 1
dial-peer voice 102 voip
destination-pattern 1111
session target ipv4:10.1.1.102
preference 1
voice class h323 1
voice class h323 1
h225 timeout tcp establish 3
Note
To simplify troubleshooting and firewall configurations, Cisco recommends that you use the new voip-gateway voip bind srcaddr command for forcing H.323 always to use a specific source IP address in call setup. Without this command, the source address that is used in the setup might vary and depends on protocol (RAS, H.225, H.245, or RTP).
Cisco VG248 Analog Phone Gateway
The Cisco VG248 Analog Phone Gateway supports the Skinny Client Control Protocol (SCCP) for clustering and failover.
Transfer calls between gateways
Using Cisco Unified Communications Manager Administration, you can configure gateways as OnNet (internal) gateways or OffNet (external) gateways by using Gateway Configuration or by setting a clusterwide service parameter. Used in conjunction with the clusterwide service parameter, Block OffNet to OffNet Transfer, the configuration determines whether calls can be transferred over a gateway.
To use the same gateway to route both OnNet and OffNet calls, associate the gateway with two different route patterns. Make one gateway OnNet and the other OffNet with both having the Allow Device Override check box unchecked.
Using
Cisco Unified Communications Manager Administration Gateway Configuration, you can
configure a gateway as OffNet or OnNet. The system considers the calls that
come to the network through that gateway OffNet or OnNet, respectively. Use the
Gateway Configuration window field, Call Classification, to configure the
gateway as OffNet, OnNet, or Use System Default. See the table below for
description of these settings.
The Route Pattern Configuration window provides a drop-down
list box called Call Classification, which allows you to configure a route
pattern as OffNet or OnNet. When Call Classification is set to OffNet and the
Allow Device Override check box is unchecked, the system considers the outgoing
calls that use this route pattern as OffNet (if configured as OnNet and check
box is unchecked, then outgoing calls are considered OnNet).
You can use the same gateway to route both OnNet and OffNet
calls by associating the gateway with two different route patterns: one OnNet
and the other OffNet, with both having the Allow Device Override check box
unchecked. For outgoing calls, the outgoing device setting classifies the call
as either OnNet or OffNet by determining whether the Allow Device Override
check box is checked.
In route pattern configuration, if the Call Classification
is set as OnNet, the Allow Device Override check box is checked, and the route
pattern is associated with an OffNet gateway, the system considers the outgoing
call OffNet.
This setting identifies the gateway as being an external
gateway. When a call comes in from a gateway that is configured as OffNet, the
outside ring gets sent to the destination device.
OnNet
This setting identifies the gateway as being an internal
gateway. When a call comes in from a gateway that is configured as OnNet, the
inside ring gets sent to the destination device.
Use System Default
This setting uses the
Cisco Unified Communications Manager clusterwide service parameter Call
Classification.
Set up transfer capabilities by using Call Classification service parameter
To configure all gateways to be OffNet (external) or OnNet
(internal), perform the following two steps:
Procedure
Step 1
Use the
Cisco Unified Communications Manager clusterwide service parameter Call
Classification.
Step 2
Configure individual gateways to Use System Default in the Call
Classification field that is on the Gateway Configuration window.
Block transfer capabilities by using service parameters
Block transfer provides a way of restricting transfer
between external devices, so fraudulent activity gets prevented. You can
configure the following devices as OnNet (internal) or OffNet (external) to
Cisco Unified Communications Manager:
H.323 gateway
MGCP FXO trunk
MGCP T1/E1 trunk
Intercluster trunk
SIP trunk
If you do not want OffNet calls to be transferred to an
external device (one that is configured as OffNet), set the
Cisco Unified Communications Manager clusterwide service parameter, Block OffNet to
OffNet Transfer, to True.
If a user tries to transfer a call on an OffNet gateway
that is configured as blocked, a message displays on the user phone to indicate
that the call cannot be transferred.
H.235 support for gateways
This feature allows
Cisco Unified Communications Manager gateways to transparently pass through the
shared secret (Diffie-Hellman key) and other H.235 data between two H.235
endpoints so that the two endpoints can establish a secure media channel.