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Table Of Contents
Release Notes for Cisco CallManager Release 3.1
IBM xSeries 340 and 330 Server Recommendations
Determining the Software Version
New Software Features in Release 3.1(1)
Survivable Remote Site Telephony (SRST) Support
Shared Resource Management Enhancements
Music on Hold Multicast and Unicast Streaming Service
T1-CAS Support in Selected VoIP Gateways
T1/E1 Primary Rate Interface (PRI) Support for Selected VoIP Gateways
Call Preservation for Selected VoIP Gateways (MGCP)
Support for Centralized Voice Messaging Application with Multiple Cisco CallManager Clusters
Cisco WebAttendant Enhancements
Cisco CallManager Redundancy for TAPI/JTAPI Applications
Overlap Sending Support through PRI ISDN
Cisco Mobile NETwork (MNET) Integration
CiscoWorks2000 Remote Syslog Analyzer Collector (RSAC) Component Not Included
Resolved Caveats - Release 3.1(1)
Open Caveats for Release 3.1(1)
Cisco IP Phone 7900 Family Administration Guide Title Changes
Remote Serviceability and Troubleshooting Information Changes Book
Maintaining Cisco IP Phone Services List
Obtaining Technical Assistance
Contacting TAC by Using the Cisco TAC Website
Release Notes for Cisco CallManager Release 3.1
August 2, 2001
These release notes describe the new features and caveats for Cisco CallManager Release 3.1.
For a list of the open and resolved caveats for Cisco CallManager Release 3.1(1), see "Resolved Caveats - Release 3.1(1)" section and "Open Caveats" section. These release notes are updated every maintenance and major release.
Use these release notes in conjunction with the Installing Cisco CallManager Release 3.1document, located on Cisco Connection Online (CCO), and the Cisco Documentation CD-ROM. The Installing Cisco CallManager Release 3.1 document comes with your CDs or convergence server.
Access the latest software upgrades and release notes for Cisco CallManager 3.1 on Cisco Connection Online (CCO) at
http://www.cisco.com/cgi-bin/tablebuild.pl/callmgr
Contents
These release notes discuss the following topics:
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Obtaining Technical Assistance
Introduction
Cisco CallManager, a network business communication system, provides high-quality telephony over IP networks. Cisco CallManager enables the conversion of conventional, proprietary, circuit-switched PBXs to multiservice, open LAN systems.
System Requirements
Make sure you install and configure Cisco CallManager Release 3.1 on a Cisco Media Convergence Server.
You may also install Cisco CallManager on a Cisco-approved Compaq server configuration or a Cisco-approved IBM server configuration.
Caution
Access the correct Cisco-approved server configuration for IBM server or Compaq server at
http://www.cisco.com/go/swonly
For system hardware component information and system requirements, refer to Installing Cisco CallManager Release 3.1.
IBM xSeries 340 and 330 Server Recommendations
Cisco recommends that if you are deploying an xSeries 340 server with a 20/40 GB DDS/4 4-mm tape drive (marketing part number for tape drive 00N7991), update your tape drive firmware to the latest version 8.160 with a release date of 2/19/01. This upgrade improves the performance of your tape drive.
Cisco recommends that if you are deploying the IBM xSeries 330 or 340 servers, update your Advanced Systems Management Processor (ASMP) firmware if necessary.
For the xSeries 340, the ASMP firmware load should be v1.15 dated 4/16/2001, and for the xSeries 330, the ASMP firmware load should be v1.04 dated 4/9/2001. The firmware upgrade ensures UM Services compatibility.
Access the correct server configuration and firmware location for IBM server or Compaq server at
http://www.cisco.com/go/swonly
Determining the Software Version
To determine the software version of Cisco CallManager 3.1, open Cisco CallManager Administration; then, click Details on the main Cisco CallManager Administration page. The following information displays:
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Compatibility Matrix
Table 1 lists minimum versions with which Cisco CallManager Release 3.1(1) has been tested. Previous versions of Cisco CallManager will not work with the versions listed below.
Related Documentation
The following list contain related documents for Cisco CallManager Release 3.1.
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New and Changed Information
The following sections contain new and changed software features for Cisco CallManager Release 3.1.
New Software Features in Release 3.1(1)
Survivable Remote Site Telephony (SRST) Support
Survivable Remote Site Telephony, a service enabled within IOS, contributes substantially to the ability of Cisco's Enterprise IP Telephony System (EIPTS) to support a single, centralized call-processing model for multiple distributed sites.
The service allows remote site IP telephony components to continue to provide service when the WAN connection between Cisco CallManagers at a central site fails. Therefore, service at a remote site survives the broken connection to the central site Cisco CallManager.
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Shared Resource Management Enhancements
Cisco CallManager Release 3.1 includes four enhancements for shared resource utilization in a cluster. These enhancements contribute substantially to the ability to support a single, centralized call-processing model for multiple distributed sites.
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Single Shared Resource per Cluster—A single shared resource may now share its services among all Cisco CallManager servers in the cluster.
In previous versions of Cisco CallManager, a single shared resource could not share its service among multiple Cisco CallManager nodes in a cluster. For example, for any cluster requiring conference bridge service, a conference bridge resource had to be configured for each Cisco CallManager in the cluster. This constraint no longer exists.
Improved Transcoder Device Efficiency—This enhancement applies to networks configured with software-only voice applications [messaging IP-interactive voice response, (IP-IVR)] that are configured to transmit and receive G.711 audio.
The Cisco CallManager using a low-bandwidth codec such as G.729 usually sets up calls placed across a low-bandwidth WAN. A call across a WAN from a Cisco IP phone to the software-only voice application therefore required a transcoder resource to transcode G.729 to G.711. Cisco CallManager 3.0 would introduce the transcoder into the Real-Time Protocol (RTP) stream as part of the call setup. However, the system was constrained in that a transcoder was inserted in the call even if the two endpoints of the call both had G.729 capabilities. This inefficient use of transcoder resources by Cisco CallManager no longer exists in Release 3.1(1). Cisco CallManager only inserts a transcoder into the call if either endpoint device does not contain a low-bandwidth codec.
Topological Association of Resources—This enhancement allows endpoint devices (Cisco IP phones, gateways, Cisco SoftPhones) to associate with locally positioned, shared-resource devices. Therefore, when a phone initiates an adhoc conference call, it can use the resources of its associated, locally positioned bridge device.
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In previous versions of Cisco CallManager, the system was unaware of the location of shared resources with respect to phones and gateways in a cluster. In a centralized call-processing environment, all transcoders and conference bridges had to be centrally located. Therefore, a conference bridge with remote site participants would require Real-Time Protocol (RTP) streaming from remote site to the central conference bridge resource, consuming costly WAN bandwidth.
Small-Site, Affordable DSP Resource Module — Prior to Cisco CallManager 3.1, Digital Signal Processor (DSP) services module products that were available for transcoding and mixed codec conference bridge service were limited to the Catalyst 6000 Voice Services Module. With the advent of the three shared resource enhancements described in this section, placement of DSP services cards at remote sites became practical. However, because the Catalyst 6000 Voice Services Card was designed and positioned for large sites, it does not produce an affordable option for these services. New, smaller scale, standalone DSP services modules will be made available shortly after Cisco CallManager 3.1 releases to more affordably provide transcoder and conference bridge services at small- to medium-sized remote sites.
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Music on Hold Multicast and Unicast Streaming Service
Music on Hold, an application that may be installed to an MCS server, streams Real-Time-Protocol (RTP) audio in either unicast or multicast streams from the application server to the endpoint device.
Administrators can stream music-on-hold audio to all IP phones, all Cisco VoIP gateways, and Cisco IP SoftPhone. Endpoint devices that support receiving multicast for music on hold include Cisco IP Phone models 7910, 7940, 7960, Cisco Catalyst 4000 Access Gateway Module (AGM) gateways, Catalyst 4224 gateways, and VG200 gateways.
A dedicated MCS server can stream as many as 250 music-on-hold streams (unicast or multicast). Any server can stream from up to fifty separate logical sources, each with its own continuously looping source .wav file. A fifty-first source - a sound card - may provide a real-time streaming source. Audio codec formats for any stream include G.711, G.729A, and high-fidelity audio (see description for this new enhancement). A translation utility included with the application allows translation from common formats such as .mp3 to the supported audio codecs.
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Issue with Music on Hold Using Locations-Based Call Admission Control
If you use locations-based call admission control, users at remote sites cannot (i.e, across a WAN link) use Music on Hold. Remote site users cannot use this feature because bandwidth calculations across locations boundries do not take into account Music on Hold streams. In place of Music on Hold, these users receive Tone on Hold, and bandwidth calculations will be correct.
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Issues with Conference Bridge Using Locations-Based Call Admission Control
Because the conference bridge resources cannot be configured with a location value, it always assumes it exists in the hub. This means that if a conference bridge is configured at a remote site (i.e., not at the hub), bandwidth would be removed from the available bandwidth, even though no bandwidth is in use across the link.
Consequently, calls may be rejected across the WAN link due to insufficient bandwidth when, in reality, enough bandwidth exists. Therefore, do not install a conference bridge at a remote site unless you can artificially adjust the bandwidth on the locations link to make more bandwidth available.
High Fidelity Audio Support
Any Cisco IP Phone 7910, 7940, and 7960 can transmit and receive audio that is sampled at 16 kHz and at a resolution of 16 bits per sample. The improved fidelity for calls through the phone handset exceeds 5-kHz audio range to the ear. This range provides a substantial improvement over a traditional 3.1 kHz audio during calls over TDM-based PBXs. As with the G.711 and G.729 codecs supported by these same phones, Cisco CallManager automatically controls the high-fidelity audio codec.
Extension Mobility Support
Extension mobility allows any user to log in to any Cisco IP Phone 7940 or Cisco IP Phone 7960. Once logged in, the user default profile, including class-of-service restrictions, primary directory number, speed dials, and productivity services apply to the phone. The user may log out manually or allow the system to log them out after timer expiration. The extension mobility application ships separately from Cisco CallManager software.
T1-CAS Support in Selected VoIP Gateways
Selected gateway interfaces including VG200, Catalyst WS-6808-T1, Catalyst 4000 AGM, Catalyst 4224, and DT-24+ gateways support T1 channel associated signaling. T1-CAS support includes E&M (ear and mouth) types 1, 2, 4, and 5.
T1/E1 Primary Rate Interface (PRI) Support for Selected VoIP Gateways
Cisco now offers T1 and E1 PRI protocols support for VG200, Catalyst 4000 AGM, and Catalyst 4224 gateways.
Call Preservation for Selected VoIP Gateways (MGCP)
Calls from any endpoint to a gateway controlled by either H.323 or Skinny Gateway Control Protocol gateways may fail when the controlling Cisco CallManager service disrupts during the call. Using Media Gateway Control Protocol (MGCP) as a substitute for H.323 or skinny gateways allows for more efficient call preservation.
In earlier versions of Cisco CallManager, MGCP support was provided for Cisco VG200 (FXS, FXO), 2600 (FXS, FXO), and 3620, 3640 and 3660-series (FXS, FXO) gateways. Two completed projects coincide with the release of Cisco CallManager 3.1 to extend MGCP support to a wider range of gateway/ time division multiplexing (TDM) interface combinations. Cisco CallManager administrative interfaces include additional MGCP configuration support for the following gateways (TDM protocols supported appear in parentheses):
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On-Hook Dialing
In Cisco CallManager 3.1, users may dial the phone without picking up the handset. The digit string initiates after the user goes offhook through the speakerphone, headset, or handset. This behavior adds to the existing dial behavior where the user dials the phone after going offhook.
Support for Centralized Voice Messaging Application with Multiple Cisco CallManager Clusters
Three enhancements to Cisco CallManager 3.1 signaling provide the support for an interface between a single voice-messaging application to multiple Cisco CallManager 3.1 clusters.
Redirected Dialed Number Identification Service (RDNIS)—Collectively, RDNIS support displays the last redirected number as well as the originally dialed number to and from configured devices and applications. While specifically designed and tested for support of voice-messaging applications, support for other applications and devices is available subject to testing and certification. Consult design guidelines for details of support.
Call Forward Number Expansion—On-net enterprise dial plans typically comprise four- or five-digit plans. Within a single site, the likelihood of digit overlap -- where two directory numbers within the same dial plan are identical -- stays minimized or eliminated because the direct inward dial (DID) number range provided by the PSTN service provider is likely from the same office and therefore nonoverlapping. However, the multisite capabilities offered by centralized call processing result in the possibility of overlap. While dial plan enhancements in Cisco CallManager 3.0 allowed operation with overlapping dial plans, delivery of dialed number information to voice-messaging systems did not accommodate overlap. Cisco CallManager 3.1 provides this overlapping dial plan support.
Call Forward Reason Code Delivery—Most voice-messaging systems can deliver prerecorded audio messages to callers. For example, mailbox owners can customize Cisco Unity mailboxes for calls forwarded on no answer, busy, or immediately on any condition. Cisco CallManager 3.1 now provides the reason code for call forwarding the voice messaging system for each call.
Serviceability Enhancements
A number of enhancements to Cisco CallManager and device instrumentation, real-time data collection and monitoring tools exist to monitor the health of the Cisco CallManager and related devices and applications. Broader instrumentation of Cisco CallManager, IP phones, gateways provides extended real-time information and alarms. Real-time Information Server (RIS), a Cisco CallManager real-time information collector is installed to Cisco CallManager nodes to collect real-time information from CCM about associated devices.
Administrator can configure the Alarm interface to deliver system events through a number of interfaces, including Event Viewer, Syslog, SDI Trace, and SDL trace, and the Alarm definitions provide system events detailed description.
An Admin Serviceability Tool (AST) comes bundled with the Cisco CallManager (but not as a standalone tool). Administrator can use this tool to monitor the health of the CCM cluster in two ways: by monitoring the vital performance counters on each CCM node in the cluster and by monitoring Device registration status and to which CCM node they are registered. The tool also provides configurable alert functionality to alert the administrator of significant events.
Cisco WebAttendant Enhancements
Cisco WebAttendant and the underlying Telephony Call Distributor have three enhancements. Longest idle hunt group logic adds to the existing linear hunt group logic. When longest idle logic is selected for a hunt group, a call sent to the hunt group goes to the available hunt group member that has been idle for the longest period. When a call passes to a Cisco WebAttendant client and the user interface is not visually focused on the display, the display will automatically "pop-to-top," regaining primary visual focus on the display. Finally, a keyboard shortcut enhancement adds function keys 'F1' through 'F4' to select among the four main windows on the Cisco WebAttendant client user interface.
Cisco CallManager Redundancy for TAPI/JTAPI Applications
In previous versions of Cisco CallManager, a TAPI or JTAPI application registered to the TAPI or JTAPI service provider on a specific Cisco CallManager node. If that Cisco CallManager failed, the application lost service and could not automatically seek an alternate TAPI or JTAPI service provider. Cisco CallManager 3.1 provides any application with the ability to automatically restore service in the event that a serving Cisco CallManager fails. A new service, the computer telephony interface (CTI) manager, is installed and runs normally on all Cisco CallManager nodes in a cluster. An application registers to a single CTI Manager for TAPI or JTAPI service. The application identifies alternate CTI Managers for CTI service. Each CTI Manager can receive call-processing services from a primary and an alternate Cisco CallManager server. Make sure applications are written to take advantage of this redundancy capability.
Overlap Sending Support through PRI ISDN
Prior versions of Cisco CallManager supported enbloc sending and receiving as well as overlap receiving through gateways with PRI ISDN protocols configured on the TDM trunk interface. Enbloc signaling sends the entire digit string to the receiving network. In the case of some PSTN switches, the switch can accept and act upon digit strings delivered in a block. However, a small percentage of PSTN switches can accept the alternative to enbloc signaling, which is overlap sending. Cisco CallManager 3.1 allows the administrator to configure any PRI ISDN trunk interface on a Cisco VG200, Cisco Catalyst 4000 AGM, Cisco Catalyst 4224, Cisco Digital Trunk Access Gateway DE-30+, or Cisco Catalyst 6608-E1 gateway configured with Euro-PRI ISDN protocol to send digits to the connected PSTN switch digit-by-digit. This overlap sending allows the attached switch to accept digits one at a time and to tell the source network (Cisco CallManager through the trunk gateway) to stop sending digits after the switch has determined that it has enough to match its dial plan. This capability, specifically developed to address the German market, may be deployed in any market where overlap sending is required.
Cisco Mobile NETwork (MNET) Integration
Cisco MNET solution comprises a composite mobile network that normally installed within an enterprise. Its initial implementation targets markets in which global system for mobile communication (GSM) cellular networks are popular. It allows an enterprise to provide extended cellular service to users when those users are located within buildings where signal strength from the public mobile cellular networks is not available due to low-signal strength. GSM cellular phones use a codec specifically designed for low-bandwidth, lower power consumption. There are two modes of this codec: GSM-FR and GSM-EFR. The FR codec provides the default codec used normally if the higher fidelity EFR codec either is not available at the other endpoint or if the codec negotiation falls back to GSM-FR. Prior to Cisco CallManager 3.1, IP phones has no access to GSM codecs, neither directly nor through transcoders. Therefore, whenever an enterprise GSM phone user wanted to talk to an IP phone when inside the enterprise buildings, the call had to be hairpinned through an AS5300 gateway for transcoding. While this allowed a connection, its implementation was costly, and the call was subject to low quality due to multiple transcoding legs. Cisco CallManager 3.1 implements GSM-FR and -EFR codecs in the Catalyst 6608-T1 and E1 Digital Signal Processor (DSP) services cards. All eight segments on these cards can be configured as transcoder resources conference bridges or trunk gateways. A call from a GSM phone may be made through a transcoder or conference bridge directly to an IP phone or directly to the WS-6608 segment configured as a trunk gateway. Cisco CallManager 3.1 administration interface supports plans for implementation in a low-cost standalone gateway at a later release.
Headset-Only Auto-Answer
You can enable or disable this feature on any line appearance of any 79XX series Cisco IP phone. When feature is enabled and a call is placed to an idle phone on which the DN on the configured line appearance has autoanswer enabled, the call automatically gets answered, and the call audio is directed to the headset audio path. If the phone is offhook while a call is made to the configured line, a visual alert lets the user know that another call is being presented to the phone. In this case, an audible alert does not occur.
Personal Directory
Personal Directory provides a personal address book stored in the Cisco CallManager LDAP directory, a Cisco IP phone synchronizer, and two Cisco IP phone services: Personal Address Book and Personal Fast Dials.
The Cisco IP Phone Address Book Synchronizer allows you to synchronize your Microsoft Outlook and/or your Outlook Express address book entries with the directory in Cisco CallManager. From a Cisco IP Phone model 7960 or 7940, you can use the Personal Address Book service to look up entries, make a selection, and press a softkey to dial the selected number.
With the Personal Fast Dials service, you can assign index numbers (from 1 to 99) for quick dialing from your Cisco IP phone. You can assign index numbers either to Personal Address Book entries or to directory entries that you add that do not correspond to the address book. You can assign and remove the Personal Fast Dials entries from your phone or the Cisco IP Phone User Options application.
CiscoWorks2000 Remote Syslog Analyzer Collector (RSAC) Component Not Included
Cisco no longer packages the CiscoWorks2000 Remote Syslog Analyzer Collector (RSAC) component with Cisco CallManager; however, the component will remain on upgrades.
With Release 3.1, you only configure the RME server in the CCM Alarm Config page. This will send ccm Syslog events directly to CiscoWorks2000 RME server, and the RSACcomponent is not needed.
For additional information, refer to the System Log Management Chapter in the Cisco CallManager Serviceability Administration Guide or CiscoWorks2000 online documentation a
http://www.cisco.com/univercd/cc/td/doc/product/rtrmgmt/cw2000/index.htm
Important Notes
Ringer Defaults to Chirp 1
When upgrading from Release 3.0(x) to Release 3.1(1), the ringer on the phone resets to Chirp 1.
You must manually reset the ringer on the Cisco IP phone to the desired state.
Resolved Caveats
Resolved Caveats - Release 3.1(1)
Table 2 lists and describes Caveats that were resolved in Cisco CallManager Release 3.1(1).
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To access the Bug Toolkit, log on to http://www.cisco.com/support/bugtools
Open Caveats
Open Caveats for Release 3.1(1)
Table 3 describes possible unexpected behaviors by Cisco CallManager Release 3.1(1). Unless otherwise noted, these caveats apply to all Cisco CallManager 3.1 releases up to and including Cisco CallManager Release 3.1(1).
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To access the Bug Toolkit, log on to http://www.cisco.com/support/bugtools.
Documentation Updates
The following section provides documentation changes that were unavailable when the Cisco CallManager Release 3.1(1) documentation suite was released.
Changes
Getting Started Title Changes
The Cisco CallManager Administration Guide and Cisco CallManager System Guide refer to the Getting Started publications provided with your phones.
Cisco IP Phone Models 7960 and 7940 User Guide replaces the Getting Started with the Cisco IP Phone 7940/7960. This document and the Getting Started with the Cisco IP Phone 7910 do not ship with the phone but are available on CCO and can be ordered.
Cisco IP Phone 7900 Family Administration Guide Title Changes
The Cisco CallManager Administration Guide and Cisco CallManager System Guide also refer to the Cisco IP Phone 7900 Family Administration Guide. This document has been renamed to Cisco IP Phone Administration Guide for Cisco CallManager.
Remote Serviceability and Troubleshooting Information Changes Book
Serviceability Administration Guide includes instructions to configure remote serviceability and to use the Cisco CallManager Trace for diagnostic traces.
Omissions
Maintaining Cisco IP Phone Services List
Using Cisco CallManager Administration, you define and maintain the list of Cisco IP Phone Services to which users can subscribe at their site. You can also create parameters for each service that require users to enter data in the Cisco IP Phone User Options application before subscribing to that service.
In the 3.1(1) release, you can mask entries in the Cisco IP Phone User Options application, so asterisks display rather than the actual user entry. You may want to do this for parameters such as passwords that you do not want others to be able to view. To mask a parameter entry, check the Parameter is a Password (mask contents) field on the Configure Cisco IP Phone Service Parameter window in CallManager Administration.
Corrections
Cisco CallManager Release 3.1(1) supports Cisco Unity Version 2.4(6.135). The Cisco CallManager System Guide incorrectly states that Cisco CallManager Release 3.1(1) requires Cisco Unity Version 3.0(1).
Obtaining Documentation
The following sections provide sources for obtaining documentation from Cisco Systems.
World Wide Web
You can access the most current Cisco documentation on the World Wide Web at the following sites:
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Documentation CD-ROM
Cisco documentation and additional literature are available in a CD-ROM package, which ships with your product. The Documentation CD-ROM is updated monthly and may be more current than printed documentation. The CD-ROM package is available as a single unit or as an annual subscription.
Ordering Documentation
Cisco documentation is available in the following ways:
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http://www.cisco.com/cgi-bin/order/order_root.pl
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http://www.cisco.com/go/subscription
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Documentation Feedback
If you are reading Cisco product documentation on the World Wide Web, you can submit technical comments electronically. Click Feedback in the toolbar and select Documentation. After you complete the form, click Submit to send it to Cisco.
You can e-mail your comments to bug-doc@cisco.com.
To submit your comments by mail, use the response card behind the front cover of your document, or write to the following address:
Attn: Document Resource Connection
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-9883We appreciate your comments.
Obtaining Technical Assistance
Cisco provides Cisco.com as a starting point for all technical assistance. Customers and partners can obtain documentation, troubleshooting tips, and sample configurations from online tools. For Cisco.com registered users, additional troubleshooting tools are available from the TAC website.
Cisco.com
Cisco.com is the foundation of a suite of interactive, networked services that provides immediate, open access to Cisco information and resources at anytime, from anywhere in the world. This highly integrated Internet application is a powerful, easy-to-use tool for doing business with Cisco.
Cisco.com provides a broad range of features and services to help customers and partners streamline business processes and improve productivity. Through Cisco.com, you can find information about Cisco and our networking solutions, services, and programs. In addition, you can resolve technical issues with online technical support, download and test software packages, and order Cisco learning materials and merchandise. Valuable online skill assessment, training, and certification programs are also available.
Customers and partners can self-register on Cisco.com to obtain additional personalized information and services. Registered users can order products, check on the status of an order, access technical support, and view benefits specific to their relationships with Cisco.
To access Cisco.com, go to the following website:
http://www.cisco.com
Technical Assistance Center
The Cisco TAC website is available to all customers who need technical assistance with a Cisco product or technology that is under warranty or covered by a maintenance contract.
Contacting TAC by Using the Cisco TAC Website
If you have a priority level 3 (P3) or priority level 4 (P4) problem, contact TAC by going to the TAC website:
http://www.cisco.com/tac
P3 and P4 level problems are defined as follows:
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In each of the above cases, use the Cisco TAC website to quickly find answers to your questions.
To register for Cisco.com, go to the following website:
http://www.cisco.com/register/
If you cannot resolve your technical issue by using the TAC online resources, Cisco.com registered users can open a case online by using the TAC Case Open tool at the following website:
http://www.cisco.com/tac/caseopen
Contacting TAC by Telephone
If you have a priority level 1 (P1) or priority level 2 (P2) problem, contact TAC by telephone and immediately open a case. To obtain a directory of toll-free numbers for your country, go to the following website:
http://www.cisco.com/warp/public/687/Directory/DirTAC.shtml
P1 and P2 level problems are defined as follows:
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This document is to be used in conjunction with the documents listed in the "Related Documentation" section.
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