Table Of Contents
New Features in Cisco IOS Release 15.0(1)XA
New Hardware Features in Cisco IOS Release 15.0(1)XA1
New Software Features in Cisco IOS Release 15.0(1)XA1
BRIDGE-MIB for Layer 2 Ethernet Switching
New Hardware Features in Cisco IOS Release 15.0(1)XA
New Software Features for Cisco IOS Release 15.0(1)XA
Unlocked Meet-me Conference Bridge
Transporting 802.1q Tags over ATM PVCs for ADSL2+
SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST
DOD MLPP PBX1 certification for CME
Call Restriction Regulations for CME sales in India
MLPP Support for Supplementary Services on SCCP Controlled Analog Endpoints
Cisco Unified Communications Manager Express 8.0 Music On Hold Enhancement (MOH)
Cisco Unified Survivable Remote Site Telephony 8.0 Music On Hold Enhancement (MOH)
RFC 4040-Based Clear-Channel Codec Negotiation for SIP Calls
Support for SIP 181 "Call is Being Forwarded" Message
Support for Expires Timer Reset for SIP 183
Support for dynamic payload type interworking for DTMF and codec packets for SIP-to-SIP calls
New Features in Cisco IOS Release 15.0(1)XA
December 21, 2009
The following features were introduced in Cisco IOS Release 15.0(1)XA. Links to feature documentation are also included.
Feature guides may contain information about more than one feature. To find information about a specific feature within a feature guide, see the Feature Information table at the end of the guide.
Feature guides document features that are supported on many different software releases and platforms. Your Cisco IOS software release or platform may not support all the features documented in a feature guide. See the Feature Information table at the end of the feature guide for information about which features in that guide are supported in your software release. Use Cisco Feature Navigator to find information about platform support and Cisco IOS and Catalyst OS software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
For the latest feature information and caveats, see the release notes for your platform and software release at:
http://www.cisco.com/en/US/products/ps6706/prod_release_notes_list.html
See the following sections for a list of features:
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New Hardware Features in Cisco IOS Release 15.0(1)XA1
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New Software Features in Cisco IOS Release 15.0(1)XA1
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New Hardware Features in Cisco IOS Release 15.0(1)XA
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New Software Features for Cisco IOS Release 15.0(1)XA
New Hardware Features in Cisco IOS Release 15.0(1)XA1
There are no new hardware features in this release.
New Software Features in Cisco IOS Release 15.0(1)XA1
BRIDGE-MIB for Layer 2 Ethernet Switching
Simple Management Network Protocol (SNMP) development and use is centered around the Management Information Base (MIB). An SNMP MIB is an abstract data base, i.e., a conceptual specification for information that a management application may read and modify in a certain form. This does not imply that the information is kept in the managed system in that same form. The SNMP agent translates between the internal data structures and formats of the managed system and the external data structures and formats defined for the MIB.
The Layer 2 Ethernet Switching Interface BRIDGE-MIB is supported in the Cisco 1861 platform from the release 15.0 (1) XA. The BRIDGE-MIB enables the user to know the Media Access Control (MAC) addresses and spanning tree information of the Ethernet switch modules. The user can query the MIB agent using the SNMP protocol and get the details of Ethernet switch modules such as MAC addresses of each interfaces and spanning protocol information.
For details to configure and retrieve the BRIDGE-MIB details, see:
http://www.cisco.com/en/US/tech/tk648/tk362/technologies_tech_note09186a0080094a9b.shtml#brgmib
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Note
The BRIDGE-MIB uses community string indexing to access a particular instance of the MIB.
New Hardware Features in Cisco IOS Release 15.0(1)XA
There are no new hardware features in this release.
New Software Features for Cisco IOS Release 15.0(1)XA
DnD Feature Button
Cisco Unified CME 8.0 adds support for the new DnD feature button. The DnD feature button allows the user to configure a line button with Do not Disturb feature instead of using the DnD softkey. Press the button to enable or disable the DnD feature when the phone is idle. This feature button can also be used as Hunt Log (HLog) button if the hunt-group logout DnD is configured in the phone. When it is used as HLog button, press the button to activate or deactivate the DnD.
Unrestricted Display
Cisco Unified CME 8.0 adds support for Unrestricted Display in ephone. This feature adds unrestricted display option for a phone type in the Cisco Unified CME system. The Unrestricted Display configuration defines a set of attributes that describe the message display option of the ephone.
Unlocked Meet-me Conference Bridge
Unlocked Meet-me conference allows the user to unlock the Meet-me conference bridge. All DN tags with the same number should be configured as unlocked. Unlocking the Meet-me conference bridge can allow unrestricted and uncontrolled access for external callers. This feature is supported only for Meet-me conference.
Transporting 802.1q Tags over ATM PVCs for ADSL2+
This feature allows 802.1q tags to be transported over Asynchronous Transfer Mode (ATM) permanent virtual circuits (PVC) used in ADSL2+ uplinks. This feature offers the following benefits:
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It allows Customer Premise Equipment (CPE) to carry traffic with a provider-specific 802.1q-tag.
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It supports the deployment of voice, video, and data services at customer premises. This service combination offers a real-time channel dedicated to Voice over IP (VoIP) traffic, and a second channel that delivers best-effort Internet service. In the current release, all traffic is marked with an 802.1p marking of 0, best-effort. This is implemented using VLAN-based service differentiation.
SIP/TLS/TCP Secure Call Signaling and SRTP Media Encryption with Cisco SRST
This feature adds Session Initiation Protocol/Transport Layer Security/Transmission Control Protocol (SIP/TLS/TCP) support for secure call signaling and Secure Real-time Transport Protocol (SRTP) for media encryption to establish a secure, encrypted connection between Cisco Unified IP Phones and a failover device using Cisco Unified Survivable Remote Site Telephony (Cisco SRST).
DOD MLPP PBX1 certification for CME
Adds enhanced Multilevel Priority and Preemption (MLPP) features for Cisco Unified CME including:
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Additional MLPP announcements for isolated code (ICA), unauthorized precedence level (UPA), loss of C2 features (LOC2), and vacant code (VCA)
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Multiple service domains for the Defense Switched Network (DSN) and Defense Red Switched Network (DRSN)
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Route codes and service digits in dialing formats
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Support for supplementary services, such as Three-Way Conferencing, Call Pickup, and Cancel Call Waiting on Analog FXS ports
Call Restriction Regulations for CME sales in India
The logical partitioning class of restriction (LPCOR) feature enables a single directory number on an IP phone or analog phone that is registered to Cisco Unified CME to connect to both PSTN and VoIP calls according to the restrictions specified by TRAI regulations. Cisco Unified CME can support both VoIP and PSTN calls while restricting the mixing of voice traffic between the PSTN and VoIP networks and preventing PSTN calls from connecting to remote locations over an IP trunk.
MLPP Support for Supplementary Services on SCCP Controlled Analog Endpoints
Adds Multilevel Priority and Preemption (MLPP) support for supplementary services, such as Three-Way Conferencing, Call Pickup, and Cancel Call Waiting on analog FXS ports. MLPP service allows validated users to place priority calls, and if necessary, to preempt lower-priority calls. The new Cancel Call Waiting feature allows a phone user to disable Call Waiting service for a call.
Cisco Unified Communications Manager Express 8.0 Music On Hold Enhancement (MOH)
Cisco Unified CME 8.0 and later versions enhances the MOH feature by playing different media streams to PSTN and VoIP callers who are placed on hold. The MOH enhancement allows you to configure up to five additional media streams supplied from multiple media files stored in a router's flash memory and eliminates the need for separate routers for streaming MOH media files.
Cisco Unified CME 8.0 MOH enhancement allows you to create MOH groups and assign ephone extension numbers to these MOH groups to receive different media streams. Callers to the extension numbers configured under the MOH groups can listen to different MOH media streams when they are placed on hold. For more information, see:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/feature/guide/MoH.html
Cisco Unified Survivable Remote Site Telephony 8.0 Music On Hold Enhancement (MOH)
Cisco Unified SRST 8.0 and later versions enhance the MOH feature by playing different media streams to PSTN and VoIP G.711 callers who are placed on hold. The MOH enhancement allows you to configure up to five additional media streams supplied from different media files stored in a router's flash memory and eliminates the need of separate routers for streaming multiple MOH media files. For more information, see:
http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/feature/guide/MOH_srst.html
CME CSTA CTI Protocol Suite
This feature in Cisco Unified CME 8.0 enables call state monitoring and control and allows third party applications to monitor and control the Cisco Unified CME system to enable programmatic control of any SCCP telephony device registered with Cisco Unified CME.
G.SHDSL Auto Pair Detect
The new dsl-group auto command automatically assigns the Central Office (CO) configuration to an ATM dsl-group on the Customer Premise Equipment (CPE) that has either Cisco HWIC-4SHDSL or Cisco HWIC-2SHDSL installed on the router. Automatic configuration is not supported on IMA groups. Automatic configuration is limited to one DSL group and ATM interface. Once a group is automatically configured, no other group can be created. All manually created groups must be deleted before creating an automatic configuration group.
Secure SIP Failover for SRST
This feature enables VoIP calls that failover to Survivable Remote Site Telephony (SRST) devices to provide secure and encrypted transport using the Session Initiation Protocol (SIP). These voice calls originate from IP Phones running the SIP protocol.
RFC 4040-Based Clear-Channel Codec Negotiation for SIP Calls
RFC4040 based clear-channel codec negotiation for SIP calls on both Cisco IOS SIP-TDM gateways and Cisco Unified Border Elements. For details about enabling this feature, see the encap clear-channel standard and voice-class sip encap clear-channel commands in the Cisco IOS Voice Command Reference:
http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html
Support for SIP 181 "Call is Being Forwarded" Message
Support for SIP 181 Call is Being Forwarded message was added to Cisco IOS SIP TDM gateways and Cisco Unified Border Elements (Cisco UBEs). This feature is enabled by default—to disable this feature for all SIP 181 messages or for SIP 181 message either with or without SDP, see the block and voice-class sip block commands in the Cisco IOS Voice Command Reference: http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html
On the Cisco UBE, this feature also adds the ability to receive SIP 181 messages on one leg and send out SIP 183 messages on the other leg. For details about enabling this feature on a Cisco UBE, see the map resp-code and voice-class sip map resp-code commands in the Cisco IOS Voice Command Reference:
http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html
Support for Expires Timer Reset for SIP 183
Support for resetting the Expires timer upon receipt of SIP 183 message so that when the terminating device lacks answer supervision or does not send the required SIP 200 OK message within the timer expiry, you can enable this feature to send periodic SIP 183 messages to reset the Expires timer and preserve the call until final response. This feature can be enabled for SIP 183 messages either with or without SDP. For details about enabling this feature, see the reset timer expires and voice-class sip reset timer expires commands in the Cisco IOS Voice Command Reference: http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html
Support for Multiple Registrars on SIP Trunks on a Cisco Unified Border Element, on Cisco IOS SIP TDM Gateways, and on Cisco Unified Communications Manager Express
Support for simultaneous registration with multiple registrars for endpoints on a Cisco Unified Border Element, a Cisco IOS SIP TDM gateway, or on Cisco Unified Communications Manager Express. This feature enhances or introduces the credentials, registrar, localhost and voice-class sip localhost commands (see the Cisco IOS Voice Command Reference at http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html) and adds support for multiple realms, which allows you to configure and enable authentication across different domains (or service providers). For details about this feature, see the Configuring Multiple Registrars on SIP Trunks feature guide: http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-multi-registrars.html
Support for PAI, PPI, and Privacy Headers on the SIP Trunk of Cisco Unified Communications Manager Express
Support for PAI, PPI, and Privacy Headers on the Cisco Unified CME SIP trunk. When enabled, Calling Number, Calling Name, and Privacy information is sent using PAI, PPI, and Privacy headers over the SIP trunk of Cisco Unified CME. This feature also enables interworking between the Remote-Party-ID (RPID) information contained in SIP line-side messages to PAI, PPI, and Privacy header information on the SIP trunk.
Support for Stripping off Progress Indicator from Incoming Q.931 CALL-PROCEEDING Messages on Cisco IOS SIP Gateways, Cisco IOS H.323 Gateways, and Cisco UBEs
Support for stripping off progress indicator (PI) from incoming Q.931 CALL-PROCEEDING message on Cisco IOS SIP and H.323 gateways and on Cisco UBEs. Configuration of this feature determines whether an incoming Q.931 CALL-PROCEEDING message with a PI value results in a SIP 183 message or H.323 Progress message. This behavior allows interworking with third-party SIP and H.323 servers. For details about enabling this feature, see the progress_ind command in the Cisco IOS Voice Command Reference at http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_book.html
IEEE 802.1ag - D8.1 standard Compliant CFM, Y.1731 multicast LBM / AIS / RDI / LCK, IP SLA for Ethernet
The ITU-Y.1731 Fault Management Functions feature adds to IEEE CFM the ETH-AIS and ETH-RDI functions for fault detection, fault verification, and fault isolation in large metropolitan-area networks (MANs) and WANs. For more information, see:
http://www.cisco.com/en/US/docs/ios/cether/configuration/guide/ce_cfm-ieee_y1731.html
Support for interworking between RSVP and non-RSVP call legs for SIP calls. Support for non-RSVP H.323 call leg to RSVP SIP call leg
The Support for Interworking Between RSVP Capable and RSVP Incapable Networks feature provides precondition-based Resource Reservation Protocol (RSVP) support for basic audio call and supplementary services on Cisco Unified Border Element (UBE). This feature improves the interoperability between RSVP and non-RSVP networks. RSVP functionality added to Cisco UBE helps you to reserve the required bandwidth before making a call.
Support for dynamic payload type interworking for DTMF and codec packets for SIP-to-SIP calls
The Support for Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls feature. This feature allows the Cisco Unified Border Element interwork between different dynamic payload type values across the call legs for the same codec. Also, Cisco UBE supports any payload type value for audio, video, named signaling events (NSEs), and named telephone events (NTEs) in the dynamic payload type range.
Support for MIB to report call volume and call rate related statistics on the Cisco Unified Border Element
Supports CISCO-VOICE-DIAL-CONTROL-MIB objects to obtain call volume and call rate information and CISCO-DSP-MGMT-MIB objects to report transcoding sessions availability information on the Cisco Unified Border Element.
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