Cisco Unified Communications SRND Based on Cisco Unified CallManager 4.x - Gateways [Cisco Unified Communications Manager (CallManager)]

Table Of Contents

Gateways

Understanding Cisco Gateways

Cisco Access Analog Gateways

Cisco Access Digital Trunk Gateways

Gateway Selection

Core Feature Requirements

Gateway Protocols

Gateway Protocols and Core Feature Requirements

DTMF Relay

Supplementary Services

Cisco Unified CallManager Redundancy

Call Survivability

Site-Specific Gateway Requirements

QSIG Support

Fax and Modem Support

Gateway Support for Fax Pass-Through and Cisco Fax Relay

Gateway Support for Modem Pass-Through

Supported Platforms and Features

Platform Protocol Support

Gateway Combinations and Interoperability of Features

Feature Support Between Similar Gateways

Gateway Configuration Examples

Cisco IOS Gateway Configuration

Cisco VG248 Configuration

Cisco Unified CallManager Configuration for Cisco IOS Gateways

Clock Sourcing for Fax and Modem Pass-Through

T.38 Fax Relay

Loose Gateway Controlled with Named Service Event (NSE)

Gateway Controlled with Capability Exchange Through H.245 or Session Description Protocol (SDP)

Call-Agent-Controlled T.38 with H.323 Annex D

Gateways for Video Telephony

Routing Inbound Calls from the PSTN

Routing Outbound Calls to the PSTN

Automated Alternate Routing (AAR)

Least-Cost Routing

ISDN B-Channel Binding, Rollover, and Busy Out

Inbound Calls

Outbound Calls

Configuring the Gateways in Cisco Unified CallManager

Call Signaling Port Numbers

Call Signaling Timers

Voice Gateways Bearer Capabilities

Gateways

Gateways provide a number of methods for connecting an IP telephony network to the Public Switched Telephone Network (PSTN), a legacy PBX, or key systems. Gateways range from specialized, entry-level and stand-alone voice gateways to high-end, feature-rich integrated router and Cisco Catalyst gateways.

This chapter explains important factors to consider when selecting a Cisco gateway to provide the appropriate protocol and feature support for your IP Telephony network. The main topics discussed in this chapter include:

•Understanding Cisco Gateways

•Gateway Selection

•QSIG Support

•Fax and Modem Support

•Gateways for Video Telephony

Understanding Cisco Gateways

Cisco access gateways allow Cisco Unified CallManager to communicate with non-IP telecommunications devices. There are two types of Cisco access gateways, analog and digital.

Cisco Access Analog Gateways

There are two categories of Cisco access analog gateways, trunk gateways and station gateways.

•Access analog station gateways

Analog station gateways connect Cisco Unified CallManager to Plain Old Telephone Service (POTS) analog telephones, interactive voice response (IVR) systems, fax machines, and voice mail systems. Station gateways provide Foreign Exchange Station (FXS) ports.

•Access analog trunk gateways

Analog trunk gateways connect Cisco Unified CallManager to PSTN central office (CO) or PBX trunks. Trunk gateways provide Foreign Exchange Office (FXO) ports for access to the PSTN, PBXs, or key systems, and E&M (recEive and transMit, or ear and mouth) ports for analog trunk connection to a legacy PBX. Whenever possible, use digital gateways to minimize any answer and disconnect supervision issues. Analog Direct Inward Dialing (DID) and Centralized Automatic Message Accounting (CAMA) are also available for PSTN connectivity.

Cisco Access Digital Trunk Gateways

A Cisco access digital trunk gateway connects Cisco Unified CallManager to the PSTN or to a PBX via digital trunks such as Primary Rate Interface (PRI), Basic Rate Interface (BRI), or T1 Channel Associated Signaling (CAS). Digital T1 PRI trunks may also be use to connect to certain legacy voice mail systems.

Gateway Selection

When selecting an IP telephony gateway, consider the following factors:

•Core Feature Requirements

•Gateway Protocols

•Gateway Protocols and Core Feature Requirements

•Site-Specific Gateway Requirements

Core Feature Requirements

Gateways used in IP telephony applications must meet the following core feature requirements:

•Dual tone multifrequency (DTMF) relay capabilities

DTMF relay capability, specifically out-of-band DTMF, separates DTMF digits from the voice stream and sends them as signaling indications through the gateway protocol (H.323, SCCP, MGCP, or SIP) signaling channel instead of as part of the voice stream or bearer traffic. Out-of-band DTMF is required when using a low bit-rate codec for voice compression because the potential exists for DTMF signal loss or distortion.

•Supplementary services support

Supplementary services are typically basic telephony functions such as hold, transfer, and conferencing.

•Fax/modem support

Fax over IP enables interoperability of traditional analog fax machines with IP telephony networks. The fax image is converted from an analog signal and is carried as digital data over the packet network. For more information, see Fax and Modem Support

•Cisco Unified CallManager redundancy support

Cisco Unified Communications is based on a distributed model for high availability. Cisco Unified CallManager clusters provide for Cisco Unified CallManager redundancy. The gateways must support the ability to "re-home" to a secondary Cisco Unified CallManager in the event that a primary Cisco Unified CallManager fails. Redundancy differs from call survivability in the event of a Cisco Unified CallManager or network failure.

Refer to the gateway product documentation to verify that any IP Telephony gateway you select for an enterprise deployment can support the preceding core requirements. Additionally, every IP Telephony implementation has its own site-specific feature requirements, such as analog or digital access, DID, and capacity requirements (see Site-Specific Gateway Requirements).

Gateway Protocols

Cisco Unified CallManager (Release 3.1 and later) supports the following gateway protocols:

•H.323

•Media Gateway Control Protocol (MGCP)

Cisco Unified CallManager Release 4.0 and later supports Session Initiation Protocol (SIP) on the trunk side. The SIP trunk implementation has been enhanced in Cisco Unified CallManager Release 5.0 to support more features.

Cisco Unified IP Phones use Skinny Client Control Protocol (SCCP), which is a lighter-weight protocol. SCCP uses a master/slave model, while H.323 is a peer-to-peer model. MGCP also follows a master/slave model.

Protocol selection depends on site-specific requirements and the installed base of equipment. For example, most remote branch locations have Cisco 2600XM, 2800, 3700, or 3800 Series routers installed. These routers support H.323 and MGCP 0.1 with Cisco IOS Release 12.2.11(T) and Cisco Unified CallManager Release 3.1 or later. For gateway configuration, MGCP might be preferred to H.323 due to simpler configuration. On the other hand, H.323 might be preferred over MGCP because of the robustness of the interfaces supported.

Simplified Message Desk Interface (SMDI) is a standard for integrating voice mail systems to PBXs or Centrex systems. Connecting to a voice mail system via SMDI and using either analog FXS or digital T1 PRI would require either SCCP or MGCP protocol because H.323 devices do not identify the specific line being used from a group of ports. Use of H.323 gateways for this purpose means the Cisco Message Interface cannot correctly correlate the SMDI information with the actual port or channel being used for an incoming call.

In addition, the Cisco Unified CallManager deployment model being used can influence gateway protocol selection. (Refer to the chapter on IP Telephony Deployment Models, page 2-1.)

Table 4-1 shows which gateways support a given protocol. Each of these protocols follows a slightly different methodology to provide support for the core gateway requirements. Gateway Protocols and Core Feature Requirements, describes how each protocol provides these feature requirements.

Table 4-1 Supported Gateway Protocols and Cisco Unified Communications Gateways

Cisco Gateway

MGCP 0.1

H.323

SCCP

SIP

Cisco 3800

Yes, beginning with Cisco IOS Release 12.3.11T

Yes, beginning with Cisco IOS Release 12.3.11T

Yes, beginning with Cisco IOS Release 12.3.11T

Yes, SIP trunk

Cisco 2800

Yes, beginning with Cisco IOS Release 12.3.8T4

Yes, beginning with Cisco IOS Release 12.3.8T4

Yes, beginning with Cisco IOS Release 12.3.8T4

Yes, SIP trunk

Cisco 3700

Yes

Supported with:

•Analog FXS/FXO

•T1 CAS (E&M Wink Start; Delay Dial only)

•T1/E1 PRI

Yes

DSP farm in Cisco IOS Release 12.2.13T

Yes, SIP trunk

Communication Media Module (CMM)

Yes

Supported with:

•T1 CAS FXS

•T1/E1 PRI

•FXS

Yes

No

Yes

Catalyst 6000
WS-X6608-x1 Gateway Module and
FXS Module WS-X6624

Yes

Supported with:

•T1 CAS E&M

•T1 CAS FXS

•T1/E1 PRI

•FXS with WS-6624

No

No

No

VG224

Yes, FXS only.

Also supports conferencing and transcoding for VG224 beginning with Cisco IOS Release 12.3(T).

Yes, FXS only

Yes, beginning with Cisco IOS Release 12.4(2)T

Yes, SIP trunk

VG248

No

No

Yes¹

No

Cisco ATA 188

Yes, FXS only

Yes, FXS only

Yes, FXS only

Yes, third-party SIP phone

Cisco AS5350

Cisco AS5400

No

Yes

No

Yes, SIP trunk

Cisco AS5850

No

Yes

No

Yes, SIP trunk

Cisco 5300

No

Yes

No

Yes, SIP trunk

Cisco 3640 and 3660

Yes

Supported with:

•Analog FXS/FXO

•T1 CAS (E&M Wink Start; Delay Dial only)

•T1/E1 PRI

Yes

DSP farm in Cisco IOS Release 12.2.13T

Yes, SIP trunk

Cisco 2600 and 2600XM²

Yes

Supported with:

•Analog FXS/FXO

•T1 CAS (E&M Wink Start; Delay Dial only)

•T1/E1 PRI

Yes

DSP farm in Cisco IOS Release 12.2.13T

Yes, SIP trunk

Cisco 1751 and 1760

Yes

Yes

Yes, conferencing and transcoding

Yes, SIP trunk

VG200³

Yes

Supported with:

•Analog FXS/FXO

•T1 CAS (E&M Wink Start; Delay Dial only)

•T1/E1 PRI

Yes

Yes (DSP farm)

No

Cisco 7200

No

Yes

No

Yes, SIP trunk

Catalyst 4000 WS-X4604-GWY Gateway Module

Yes

Yes

No

No

Cisco ICS7750-MRP

No

Yes

No

No

Cisco ICS7750-ASI

No

Yes

No

No

DE-30+, DT-24+⁴

Yes

No

No

No

Cisco 827-V44

No

Yes, supported for FXS

No

No

¹The VG248 is not a true gateway in that it uses Skinny Client Control Protocol (SCCP) instead of H.323, MGCP, or SIP.

²For IP Telephony applications, use Cisco 2800 Series Routers. For memory considerations for the Cisco 2600 routers, see the Product Bulletin at /en/US/products/hw/routers/ps259/prod_bulletin09186a0080088755.html

³The VG200 is no longer available for purchase and has been replaced by the Cisco 2800 Router. Existing models of the VG200 can still be used in an IP Telephony deployment.

⁴These models have reached end of life.

Note Prior to deployment, check the Cisco IOS software release notes to confirm feature or interface support.

Gateway Protocols and Core Feature Requirements

This section describes how each protocol (SCCP, H.323, MGCP, and SIP) supports the following gateway feature requirements:

•DTMF Relay

•Supplementary Services

•Cisco Unified CallManager Redundancy

DTMF Relay

Dual-Tone Multifrequency (DTMF) is a signaling method that uses specific pairs of frequencies within the voice band for signals. A 64 kbps pulse code modulation (PCM) voice channel can carry these signals without difficulty. However, when using a low bite-rate codec for voice compression, the potential exists for DTMF signal loss or distortion. An out-of-band signaling method for carrying DTMF tones across a Voice over IP (VoIP) infrastructure provides an elegant solution for these codec-induced symptoms.

SCCP Gateways

The SCCP gateways, such as the Cisco VG248, carry DTMF signals out-of-band using Transmission Control Protocol (TCP) port 2002. Out-of-band DTMF is the default gateway configuration mode for the VG248.

H.323 Gateways

The H.323 gateways, such as the Cisco 3700 series products, can communicate with Cisco Unified CallManager using the enhanced H.245 capability for exchanging DTMF signals out-of-band. The following is an example out-of-band DTMF configuration on a Cisco IOS gateway:
dial-peer voice 100 voip
destination-pattern 555....
session target ipv4:10.1.1.1
CODEC g729ar8
dtmf-relay h245-alphanumeric
preference 0
MGCP Gateway

The Cisco IOS-based VG224, 2600XM, 2800, 3700, and 3800 platforms use MGCP for Cisco Unified CallManager communication. Within the MGCP protocol is the concept of packages. The MGCP gateway loads the DTMF package upon start-up. The MGCP gateway sends symbols over the control channel to represent any DTMF tones it receives. Cisco Unified CallManager then interprets these signals and passes on the DTMF signals, out-of-band, to the signaling endpoint. The global configuration command for DTMF relay is:
mgcp dtmf-relay CODEC all mode out-of-band
You must enter additional configuration parameters in the Cisco Unified CallManager MGCP gateway configuration interface.

The Catalyst 6000, DE-30+, and DT-24+ all support MGCP with Cisco Unified CallManager Release 3.1 and later. DTMF relay is enabled by default and does not need additional configuration.

SIP Gateway

The Cisco IOS-based VG224, 2600XM, 2800, 3700, 3800 platforms can use SIP for Cisco Unified CallManager communication. They support various methods for DTMF, but only the following two methods can be used to communicate with Cisco Unified CallManager:

•Named Telephony Events (NTE), or RFC 2833

•Unsolicited SIP Notify (UN)

The following example shows a configuration for NTE:
dial-peer voice 100 voip
destination-pattern 555....
session target ipv4:10.1.1.1
session protocol sipv2
dtmf-relay rtp-nte
The following example shows a configuration for UN:
dial-peer voice 100 voip
destination-pattern 555....
session target ipv4:10.1.1.1
session protocol sipv2
dtmf-relay sip-notify
For more details on DTMF method selection, see the chapter on Media Resources, page 6-1.

Supplementary Services

Supplementary services provide user functions such as hold, transfer, and conferencing. These are considered fundamental requirements of any voice installation. Each gateway evaluated for use in an IP telephony network should provide support for supplementary services natively, without the use of a software media termination point (MTP).

SCCP Gateways

The Cisco VG224, VG248, and ATA 188 gateways provide full supplementary service support. The SCCP gateways use the Gateway-to-Cisco Unified CallManager signaling channel and SCCP to exchange call control parameters.

H.323 Gateways

H.323v2 implements Open/Close LogicalChannel and the emptyCapabilitySet features. The use of H.323v2 by H.323 gateways, beginning in Cisco IOS Release 12.0(7)T and Cisco Unified CallManager Release 3.0 and later, eliminates the requirement for an MTP to provide supplementary services. With Cisco Unified CallManager Release 3.1 and later, a transcoder is allocated dynamically only if required during a call to provide access to G.711-only devices while still maintaining a G.729 stream across the WAN. Full support for H.323v2 is available in Cisco IOS Release 12.1.1T.

Once an H.323v2 call is set up between a Cisco IOS gateway and an IP phone, using the Cisco Unified CallManager as an H.323 proxy, the IP phone can request to modify the bearer connection. Because the Real-Time Transport Protocol (RTP) stream is directly connected to the IP phone from the Cisco IOS gateway, a supported voice codec can be negotiated.

Figure 4-1 and the following steps illustrate a call transfer between two IP phones:

1. If IP Phone 1 wishes to transfer the call from the Cisco IOS gateway to Phone 2, it issues a transfer request to Cisco Unified CallManager using SCCP.

2. Cisco Unified CallManager translates this request into an H.323v2 CloseLogicalChannel request to the Cisco IOS gateway for the appropriate SessionID.

3. The Cisco IOS gateway closes the RTP channel to Phone 1.

4. Cisco Unified CallManager issues a request to Phone 2, using SCCP, to set up an RTP connection to the Cisco IOS gateway. At the same time, Cisco Unified CallManager issues an OpenLogicalChannel request to the Cisco IOS gateway with the new destination parameters, but using the same SessionID.

5. After the Cisco IOS gateway acknowledges the request, an RTP voice bearer channel is established between Phone 2 and the Cisco IOS gateway.

Figure 4-1 H.323 Gateway Supplementary Service Support

MGCP Gateway

The MGCP gateways provide full support for the hold, transfer, and conference features through the MGCP protocol. Because MGCP is a master/slave protocol with Cisco Unified CallManager controlling all session intelligence, Cisco Unified CallManager can easily manipulate MGCP gateway voice connections. If an IP telephony endpoint (for example, an IP phone) needs to modify the session (for example, transfer the call to another endpoint), the endpoint would notify Cisco Unified CallManager using SCCP. Cisco Unified CallManager then informs the MGCP gateway, using the MGCP User Datagram Protocol (UDP) control connection, to terminate the current RTP stream associated with the Session ID and to start a new media session with the new endpoint information. Figure 4-2 illustrates the protocols exchanged between the MGCP gateway, endpoints, and Cisco Unified CallManager.

Figure 4-2 MGCP Gateway Supplementary Service Support

SIP Gateway

The Cisco Unified CallManager SIP trunk interface to Cisco IOS SIP gateways supports supplementary services such as hold, blind transfer, and attended transfer. The support for supplementary services is achieved via SIP methods such as INVITE and REFER. For more details, refer to the following documentation:

•Cisco Unified CallManager 5.0 System Guide, available at

http://www.cisco.com

•Cisco IOS SIP Configuration Guide, available at

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a00807517b8.html

Cisco Unified CallManager Redundancy

An integral piece of the IP telephony architecture is the provisioning of low-cost, distributed PC-based systems to replace expensive and proprietary legacy PBX systems. This distributed design lends itself to the robust fault tolerant architecture of clustered Cisco Unified CallManagers. Even in its most simplistic form (a two-system cluster), a secondary Cisco Unified CallManager should be able to pick up control of all gateways initially managed by the primary Cisco Unified CallManager.

SCCP Gateways

Upon boot-up, the Cisco VG224, VG248, and ATA 188 gateways are provisioned with Cisco Unified CallManager server information. When these gateways initialize, a list of Cisco Unified CallManagers is downloaded to the gateways. This list is prioritized into a primary Cisco Unified CallManager and secondary Cisco Unified CallManager. In the event that the primary Cisco Unified CallManager becomes unreachable, the gateway registers with the secondary Cisco Unified CallManager.

H.323 Gateways

Using several enhancements to the dial-peer and voice class command sets in Cisco IOS Release 12.1(2)T, Cisco H.323 gateways support redundant Cisco Unified CallManagers. A new command, H.225 tcp timeout <seconds>, has been added. This command tracks the time it takes for the H.323 gateway to establish an H.225 control connection for H.323 call setup. If the H.323 gateway cannot establish an H.225 connection to the primary Cisco Unified CallManager, it tries a second Cisco Unified CallManager defined in another dial-peer statement. The H.323 gateway shifts to the dial-peer statement with next highest preference setting. The following commands allow you to configure Cisco Unified CallManager redundancy for a H.323 gateway:
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 tcp timeout <1-30 sec>
MGCP Gateway

MGCP gateways also have the ability to fail over to a secondary Cisco Unified CallManager in the event of communication loss with the primary Cisco Unified CallManager. When the failover occurs, active calls are preserved.

Within the MGCP gateway configuration file, the primary Cisco Unified CallManager is identified using the call-agent <hostname> command, and a list of secondary Cisco Unified CallManager is added using the ccm-manager redundant-host command. Keepalives with the primary Cisco Unified CallManager are through the MGCP application-level keepalive mechanism, whereby the MGCP gateway sends an empty MGCP notify (NTFY) message to Cisco Unified CallManager and waits for an acknowledgement. Keepalive with the backup Cisco Unified CallManagers is through the TCP keepalive mechanism.

If the primary Cisco Unified CallManager becomes available at a later time, the MGCP gateway can "re-home," or switch back to the original Cisco Unified CallManager. This re-homing can occur either immediately, after a configurable amount of time, or only when all connected sessions have been released. This is enabled through the following global configuration commands:
ccm-manager redundant-host <hostname1 | ipaddress1 > <hostname2 | ipaddress2>
[no] call-manager redundancy switchback [immediate|graceful|delay <delay_time>]
SIP Gateway

Redundancy with Cisco IOS SIP gateways can be achieved similarly to H.323. If the SIP gateway cannot establish a connection to the primary Cisco Unified CallManager, it tries a second Cisco Unified CallManager defined under another dial-peer statement with a higher preference.

By default the Cisco IOS SIP gateway transmits the SIP INVITE request 6 times to the Cisco Unified CallManager IP address configured under the dial-peer. If the SIP gateway does not receive a response from that Cisco Unified CallManager, it will try to contact the Cisco Unified CallManager configured under the other dial-peer with a higher preference.

Cisco IOS SIP gateways wait for the SIP 100 response to an INVITE for a period of 500 ms. By default, it can take up to 3 seconds for the Cisco IOS SIP gateway to reach the backup Cisco Unified CallManager. You can change the SIP INVITE retry attempts under the sip-ua configuration by using the command retry invite <number>. You can also change the period that the Cisco IOS SIP gateway waits for a SIP 100 response to a SIP INVITE request by using the command timers trying <time> under the sip-ua configuration.

One other way to speed up the failover to the backup Cisco Unified CallManager is to configure the command monitor probe icmp-ping under the dial-peer statement. If Cisco Unified CallManager does not respond to an Internet Control Message Protocol (ICMP) echo message (ping), the dial-peer will be shut down. This command is useful only when the Cisco Unified CallManager is not reachable. ICMP echo messages are sent every 10 seconds.

The following commands enable you to configure Cisco Unified CallManager redundancy on a Cisco IOS SIP gateway:
sip-ua
 retry invite <number>
 timers trying <time>
dial-peer voice 101 voip
 destination-pattern 2...
 session target ipv4:10.1.1.101
 preference 0
 monitor probe icmp-ping
 session protocol sipv2
dial-peer voice 102 voip
 destination-pattern 2...
 session target ipv4:10.1.1.102
 preference 1
 monitor probe icmp-ping
 session protocol sipv2
Call Survivability

Prior to Cisco Unified CallManager 4.2, call survivability was available only with MGCP gateways. If the signaling component of a call disappeared, the media connection was preserved until call termination, thereby allowing completion of the call.

Cisco Unified CallManager 4.2 introduces a new H.323 feature called quiet clear. In addition, Cisco IOS Release 12.4(4)XC introduces H.323 VoIP call preservation enhancements for WAN link failures. Both Cisco Unified CallManager and the H.323 gateway must be configured appropriately in order to allow call survivability.

For details on configuring the H.323 gateway, refer to the Cisco IOS H.323 Configuration Guide, available at

http://www.cisco.com

Site-Specific Gateway Requirements

Each IP Telephony implementation has its own site-specific requirements. The following questions can help you with IP Telephony gateway selection:

•Is the PSTN (or PBX) access analog or digital?

•What type of analog (FXO, FXS, E&M, DID, CAMA) or digital (T1, E1, CAS, CCS) interface is required for the PSTN or PBX?

•If the PSTN access is digital, what type of signaling is required (T1 CAS, Q.931 PRI, E1 CAS, or R2)?

•What type of signaling does the PBX currently use?

–FXO or FXS: loop start or ground start

–E&M: wink-start, delay-start, or immediate-start

–E&M: type I, II, III, IV, or V

–T1: CAS, Q.931 PRI (User-Side or Network-Side), QSIG, DPNSS, or Proprietary d-channel (CCS) signaling

–E1: CAS, R2, Q.931 PRI (User-Side or Network-Side), QSIG, DPNSS, Proprietary d-channel (CCS) signaling

•What type of framing (SF, ESF, or G.704) and line encoding (B8ZS, AMI, CRC-4, or HDB3) does the PBX currently use?

•Does the PBX require passing proprietary signaling? If so, which time slot is the signaling passed on, and is it HDLC-framed?

•What is the required capacity of the gateway; that is, how many channels are required? (Typically, if 12 or more voice channels are required, then digital is more cost effective than an analog solution.)

•Is Direct Inward Dialing (DID) required? If so, specify analog or digital.

•Is Calling Line ID (CLID) needed?

•Is Calling Name needed?

•What types of fax and modem support are required?

•What types of voice compression are required?

•What types of supplementary services are required?

•Will the PBX provide clocking, or will it expect the Cisco gateway to provide clocking?

•Is rack space available for all needed gateways, routers, and switches?

Note Direct Inward Dial (DID) refers to a private branch exchange (PBX) or Centrex feature that permits external calls to be placed directly to a station line without use of an operator.

Note Calling Line Identification (CLI, CLID, or ANI) refers to a service available on digital phone networks to display the calling number to the called party. The central office equipment identifies the phone number of the caller, enabling information about the caller to be sent along with the call itself. CLID is synonymous with Automatic Number Identification (ANI).

Cisco Unified Communications gateways are able to inter-operate with most major PBX vendors, and they are EIA/TIA-464B compliant.

The site-specific and core gateway requirements are a good start to help narrow the possible choices. Once you have defined the required features, you can make a gateway selection for each of the pertinent configurations, whether they are single-site enterprise deployments of various sizes and complexities or multisite enterprise deployments.

The following tables summarize the features and interface types supported by the various Cisco gateway models.

Note In the following tables, the Cisco IOS and Cisco Unified CallManager release numbers refer to the minimum release that can support the listed feature on a particular gateway platform. For specific recommendations about the preferred software release for each hardware platform, refer to the documentation at
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_software_versions_comparison.html

Cisco Analog Gateways

Table 4-2 lists supported interface types for Cisco analog gateways using H.323 or Session Initiation Protocol (SIP), and Table 4-3 lists supported interface types for Cisco analog gateways using Media Gateway Control Protocol (MGCP).

Table 4-2 Supported Analog H.323 and SIP Features

Cisco Gateway

Interface Type

FXS

FXO

E&M

FXO, Battery Reversal

Analog DID

CAMA 911

3800 Series

Yes

Yes

Yes

Yes

Yes

Yes

2800 Series

Yes

Yes

Yes

Yes

Yes

Yes

3700 Series

Yes

Yes

Yes

Yes

Yes

Yes

Communication Media Module (CMM) 24FXS

Yes

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

N/A

N/A

N/A

N/A

N/A

N/A

6608 and 6624

N/A

N/A

N/A

N/A

N/A

N/A

VG224

Yes

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

Analog Telephone Adapter (ATA)

Yes

No

No

No

No

No

3600 Series

Yes

Yes

Yes

Yes

Yes

12.2.11T

2600 Series

Yes

Yes

Yes

Yes

Yes

12.2.11T

1751 and 1760

Yes

Yes

Yes

Yes

Yes

Yes

VG200

Yes

Yes

Yes

No

Yes

No

7x00 family

N/A

N/A

N/A

N/A

N/A

N/A

ICS 7750

Yes

Yes

Yes

Yes

Yes

No

Catalyst 4000 Access Gateway Module (AGM)

Yes

Yes

No

No

No

No

827-4V¹

Yes

No

No

No

No

No

¹This model has reached end of life.

Table 4-3 Supported Analog MGCP Features

Cisco Gateway

Interface Type

FXS

FXO

E&M

FXO, Battery Reversal

Analog DID

CAMA 911

3800 Series

Yes

Yes

No

Yes

No

No

2800 Series

Yes

Yes

No

Yes

No

No

3700 Series

Yes

Yes

No

Yes

No

No

Communication Media Module (CMM) 24FXS

Yes

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

N/A

N/A

N/A

N/A

N/A

N/A

6608 and 6624

Yes

No

No

No

No

No

VG224

Yes

No

No

No

No

No

VG248

No

No

No

No

No

No

Analog Telephone Adapter (ATA)

Yes

N/A

N/A

N/A

N/A

N/A

3600 Series

Yes

Yes

No

Yes

No

No

2600 Series

Yes

Yes

No

Yes

No

No

1751 and 1760

Yes

Yes

No

Yes

No

No

VG200

Yes

Yes

No

Yes

No

No

7x00 family

N/A

N/A

N/A

N/A

N/A

N/A

ICS 7750

Yes

Yes

No

No

No

No

Catalyst 4000 Access Gateway Module (AGM)

Yes

Yes

No

No

No

No

827-4V¹

No

No

N/A

N/A

N/A

N/A

¹This model has reached end of life.

Cisco Digital Gateways

Table 4-4 through Table 4-7 list supported interface types for Cisco digital gateways using H.323 or Session Initiation Protocol (SIP). Table 4-8 lists supported interface types for Cisco digital gateways using Media Gateway Control Protocol (MGCP).

Table 4-4 Supported Digital H.323 and SIP Features for BRI, T1 CAS, T1 FGB, T1 FGD, and T1 QSIG

Cisco Gateway

Interface Type

BRI (TE, User side)

BRI (NT, Network side)

BRI QSIG (Net3)

BRI Phones

T1 CAS (Robbed bit)

T1 FGB

T1 FGD

T1 QSIG

3800 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

2800 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

3700 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

Communication Media Module (CMM) 24FXS

N/A

N/A

N/A

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

N/A

N/A

N/A

N/A

Yes

No

No

Yes

6608 and 6624

N/A

N/A

N/A

N/A

No

No

No

No

VG224

N/A

N/A

N/A

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

No

No

Analog Telephone Adapter (ATA)

No

No

No

No

No

No

No

No

3600 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

2600 Series

Yes

Yes

Yes

No

Yes

No

Yes

Yes

1751 and 1760

No

Yes

Yes

No

Yes

No

No

Yes

VG200

Yes

Yes

No

No

Yes

No

Yes

No

7x00 family

N/A

N/A

N/A

N/A

Yes

No

Yes

Yes

ICS 7750

Yes

Yes

No

No

Yes

No

Yes

No

Catalyst 4000 Access Gateway Module (AGM)

Yes

No

Yes

No

Yes

No

Yes

Yes

827-4V¹

No

No

No

No

No

No

No

No

¹This model has reached end of life.

Table 4-5 Supported Digital H.323 and SIP Features for T1 PRI SL-1, 4ESS, and 5ESS

Cisco Gateway

Interface Type

T1 PRI (User, DMS-100)

T1 PRI (Network, SL-1)

T1 PRI (User, 4ESS)

T1 PRI (Network, 4ESS)

T1 PRI (User, 5ESS)

T1 PRI (Network, 5ESS)

3800 Series

Yes

Future

Yes

Yes

Yes

Yes

2800 Series

Yes

Future

Yes

Yes

Yes

Yes

3700 Series

Yes

Future

Yes

Future

Yes

Future

Communication Media Module (CMM) 24FXS

N/A

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

Yes

Yes

Yes

Yes

Yes

Yes

6608 and 6624

No

No

No

No

No

No

VG224

N/A

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

Analog Telephone Adapter (ATA)

No

No

No

No

No

No

3600 Series

Yes

Future

Yes

Yes

Yes

Yes

2600 Series

Yes

Future

Yes

Yes

Yes

Yes

1751 and 1760

Yes

Future

Yes

Future

Yes

Future

VG200

Yes

No

Yes

No

Yes

No

7x00 family

Yes

Future

Yes

Future

Yes

Future

ICS 7750

Yes

No

Yes

No

Yes

No

Catalyst 4000 Access Gateway Module (AGM)

Yes

Future

Yes

Future

Yes

Future

827-4V¹

No

No

No

No

No

No

¹This model has reached end of life.

Table 4-6 Supported Digital H.323 and SIP Features for T1 PRI NI2, NFAS, and Network Specific Facilities (NSF) Service

Cisco Gateway

Interface Type

T1 PRI (User, NI2)

T1 PRI (Network, NI2)

T1 PRI NFAS (User, DMS-100)

T1 PRI NFAS (User, 4ESS)

T1 PRI NFAS (User, 5ESS)

T1 PRI (Megacom or SDN, 4ESS)

3800 Series

Yes

Yes

Yes

Yes

Yes

Yes

2800 Series

Yes

Yes

Yes

Yes

Yes

Yes

3700 Series

Yes

Yes

Yes

Yes

Yes

Yes

Communication Media Module (CMM) 24FXS

N/A

N/A

N/A

N/A

N/A

N/A

CMM-6T1/E1

Yes

Yes

Yes

Future

Future

No

6608 and 6624

No

No

No

No

No

No

VG224

N/A

N/A

N/A

N/A

N/A

N/A

VG248

No

No

No

No

No

No

Analog Telephone Adapter (ATA)

No

No

No

No

No

No

3600 Series

Yes

Yes

Yes

Yes

Yes

Yes

2600 Series

Yes

Yes

Yes

Yes

Yes

Yes

1751 and 1760

Yes

Yes

No

No

No

No

VG200

Yes

Yes

No

No

No

No

7x00 family

Yes

Yes

No

No

No

No