CCNP Voice TVoice 642-427 Quick Referenceebook.konfigurasi.net/Cisco/CCNP-Voice-Tvoice.pdf · This Quick Reference is a final preparation resource. With the assistance of this guide

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From the Library of Vittorio Valentini

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CCNP Voice TVoice 642-427 Quick Reference

Table of Contents

Section 1 Troubleshooting Cisco Unified Communications v8.0 ...............................................4

Section 2 Real Time Monitoring Tool ...........................7

Section 3 Cisco Unified Serviceability .....................15

Section 4 Command-Line Interface ............................33

Section 5 Troubleshooting Endpoint Registration .....................................................................................36

Section 6 MGCP ........................................................................39

Section 7 H.323 and SIP Gateways ............................44

Section 8 Troubleshooting Database Issues ........49

Section 9 Troubleshooting Call Setup Issues ......54

Section 10 Tracing Calls .....................................................63

Section 11 Troubleshooting Off-Net Call Setup .........................................................................................65

Section 12 Troubleshooting Voice Quality Issues ................................................................................66

Section 13 Cisco Service Advertisement Framework ..................................................70

Section 14 Troubleshooting Device Mobility Issues ..............................................................................79

Section 15 Extension Mobility Issues ........................82

Section 16 Cisco Unified Mobility General Issues ..............................................................................86

Section 17 Troubleshooting Media Resources ........................................................................................89

Section 18 Troubleshooting Media Termination Points .....................................................................92

Section 19 Troubleshooting Conferences .............94

Section 20 Troubleshooting Transcoder Issues......................................................................97

Brion S. Washington

ciscopress.com


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© 2011 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 99 for more details.


About the AuthorBrion S. Washington, CCNA, is a senior voice engineer consultant in Atlanta, GA. He has more than ten years of Cisco experience, with the last five years dedicated to VoIP; he has worked with all the Cisco VoIP products. Brion has done many large projects involving VoIP, from complete network design, implementation, and the last level of escalation. He is currently finishing up his CCVP.

About the Technical ReviewerAlex Hannah, CCIE Voice No. 25853, is a certified Cisco instructor, specializing in teaching the Cisco Advanced IP Communications product line. He has more than seven years consulting experience in Cisco Unified Communications for SMB through Enterprise spaces. He is president of Hannah Technologies LLC, a Richmond, Virginia-based Cisco consulting firm specializing in Cisco Advanced IP Communications and application development using Microsoft technologies. He holds a bachelor’s degree in Information System from Virginia Commonwealth University with a minor in Business. Additionally, he is the founder of UCCX.net, a video-based training website for the Cisco UC product line. In his spare time, you can find Alex on his boat wakeboarding with his family and friends.


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DedicationsTons of people were a part of making this guide what it is. I personally want to thank each of them.

My family: My wife Christi, and kids Taylor, Brittney, and Nathan for their patience and understanding during this project. They kept me sane.

Alex Hannah: Whose website I visited before while trying to understand UCCX. His meticulous technical editing and knowledge guided this project from infancy to maturity. Thank you, Alex.

Amanda Finley: For the hours we spent playing Bad Company 2 on the PS3, thanks for helping me relax and enjoy my down time.

The Cisco Press Team: Brett Bartow, Mandie Frank, Keith Cline, and Sheri Cain. They are the biggest reason this project was successful. I appreciate all that the team did for me and the project.

Brion S.Washington


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Section 1

Troubleshooting Cisco Unified Communications v8.0

The Troubleshooting Cisco Unified Communications v8.0 exam tests your ability to isolate problems and use an assortment of tools and troubleshooting methodologies. This exam also tests your understanding of how components work and communicate with each other. No troubleshooting is “by the numbers” per se, and problems can range from very simple to never before seen.

By understanding how to best approach a problem and what tools will help isolate the problem, you are more likely to resolve the issue in a timely manner the first time.

This Quick Reference guide gives you quick facts about the best way to approach a problem, the tools available, and additional areas that Cisco might choose to include on the exam. This Quick Reference is a final preparation resource. With the assistance of this guide and other study aids, you will build a solid repertoire of skills that will help prepare for the TVOICE exam and any real-world issues you might face.

Cisco Way of TroubleshootingOver the years, Cisco has come up the idea of a structured approach to resolving issues on either data or voice networks. A structured approach breaks the process down into easily implemented tasks to more quickly resolve problems. The idea behind this methodology is to assist in the quick and accurate resolution of any issue that can cause unwanted network behavior. The process is simple after you work through a few of them. This eight-step process has helped numerous people. These steps produce a baseline of the issue you are experiencing and a practical and structured approach to problem solving:

1. Define the problem. Clearly state what the problem is. List the symptoms and their causes.

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Section 1 Troubleshooting Cisco Unified Communications v8.0

2. Gather the facts. Get information from the affected users and technical staff to help isolate the issue. Use tools such as trace files, logs, router diagnostics output, and so on to get more information.

3. Consider possible causes. Start with the more likely causes. This step should help eliminate some of the choices.4. Create an action plan. Your plan should begin with the most likely cause. The plan should only change one variable at a

time. It is crucial to change or troubleshoot only one area at a time to rule out a potential resolution.5. Implement the action plan. Perform your action plan one action at a time. Watch carefully to see whether the desired results

have been accomplished.6. Check the results. Check to see whether problem was resolved. Use the same tools used to find the problem.7. Analyze the results. Has the issue been resolved? If yes, go to Step 8. If no, go back to Step 4.8. Congratulate yourself. The process is complete.

Example 1A user calls and says she cannot complete calls to directory number (DN) 4005.

Here is the structured approach at work:

1. The user cannot call DN 4005.2. The call fails to DN 4005. Other users cannot call DN 4005 either. Check with the network administrator and verify the network is running okay.3. Configuration issue, network issue, bad phone.4. Check Communication Manager to see whether the directory number is used.5. Communication Manager was checked, and the phone is no longer issued to a user or phone.6. The problem was resolved, and users were informed that is DN is no longer valid.7. N/A.8. Document this resolution.

That is a basic example of structured troubleshooting at work.


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Section 1 Troubleshooting Cisco Unified Communications v8.0

Tools of the TradeTo perform any job, you need the correct tools and to know how to use them. Cisco Unified Communications has a few built-in tools that, along with other external tools, can help in your problem-resolution process.


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Section 2

Real Time Monitoring Tool

Cisco Unified Real Time Monitoring Tool (RTMT) is a real-time information management tool that you can use on Cisco Communication Managers, IP phones, CTI, and so on. Figure 2-1 shows an image of the tool and several of its useful features.

Figure 2-1 Shows the Main RTMT Page


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Section 2 Real Time Monitoring Tool

RTMT enables real-time monitoring of Cisco Unified Communication Managers (CUCM) and devices, and

■ Includes performance counters.

■ Monitors client-side applications.

■ Trace and log files are accessible through this tool.

■ Uses HTTPS and TCP to communicate with CUCM.

■ Includes a comprehensive help section.

System SummaryThis provides information about the current state of the system. The page displays virtual memory usage, CPU usage, alert history, and partition information.

Server SectionVirtual Memory Usage and CPU UsageThis section provides detailed information about the memory available and CPU usage. If you suspect CPU problems or memory issues, look here first.

ProcessThe Process page lists all running processes on the server you have selected:

■ PID (process ID)

■ State of the process (hung and so on)

■ % of CPU usage

■ Shared memory

■ Page count


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Disk UsageThe Disk Usage page, displayed in Figure 2-2, shows disk usage in megabytes. Look here if you are experiencing disk issues:

■ Common partition usage

■ Swap partition usage

■ Spare partition usage

FigURE 2-2 Shows the Disk Usage


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Critical ServicesCisco attempted to make gathering information easy when it designed this page. Figure 2-3 shows the Critical Services page, where you can view all the services and features that run on the server, cluster, or system:

■ State of services

■ Status of services

■ Elapsed time/not applicable

FigURE 2-3 Example of Critical Services


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To start a nonrunning service, follow these steps:1. Right-click the service and click Control Center.2. Select the server you need.3. Select the feature.4. Save the changes.

Trace and Log CentralTrace and Log Central provides the only location in a CUCM server to gather log files. You can pull the Signal Distribution Layer (SDL) and System Diagnostic Interface (SDI) trace files from any server in the cluster with this tool. It proves invaluable for not only troubleshooting, but also opening up dreaded Technical Assistance Center (TAC) cases. Figure 2-4 shows the status of collecting files.

FigURE 2-4 Process Gathering Page


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Alert CentralAlert Central shows the status and history of every alert in the cluster. There are two types of alerts: predefined and user defined. The predefined alerts cannot be deleted. User-defined alerts can generate emails to administrators in response to alerts, a helpful feature. Figure 2-5 gives an idea of the information contained in Alert Central.

FigURE 2-5 Alert Central


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The Alert menu consists of the following:■ Set alerts: Allows alerts and alert properties to be set■ Remove alerts: Removes alerts■ Enable alerts: Enables alerts■ Disable alerts: Disables alerts■ Suspend cluster/Node alerts: Temporarily suspends alerts on the Communication Manager or the cluster■ Clear alerts: Clears the alert that has been checked. Color changes from red to black■ Clear all alerts: Clears all alerts■ Alert details: Provides details of the alert

Syslog ViewerThis useful tool finds problems with the Unified Communications Manager. You can locate issues that cause system process hangs/crashes. Figure 2-6 shows areas that can assist you in locating and resolving issues.


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FigURE 2-6 Syslog Viewer Page


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Section 3

Cisco Unified Serviceability

A main section of CUCM is Unified Serviceability. Within in this section, you have access two useful web-based tools and resources that can be crucial to a healthy Unified Communications environment: Alarm Configuration and Traces. We discuss both here.

Alarm ConfigurationThe options for configuring Alarms are clearly shown in Figure 3-1. You must select the server and service, and then chose which alarms you want to set up:

■ Configuration (where you configure your alarms)

■ Definitions (provides alarms definitions)

■ Select server

■ Select service

■ Select service group


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Section 3 Cisco Unified Serviceability

Alarm destinations are as follows:

■ Local syslogs

■ Remote syslogs

■ SDI (System Diagnostic Interface)

■ SDL (Signal Distribution Layer)

FigURE 3-1 Alarm Configuration Page


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All alarms have a few options in the drop-down list:

■ Emergency: Designates the system is unusable.

■ Alert: Immediate action is required.

■ Critical: Critical issue.

■ Error: An error exists.

■ Warning: Informational warning message.

■ Notice: Significant condition, but normal.

■ Informational: Informational messages.

■ Debug: TAC engineers use these for debugging.

TracesThis section is where you configure the traces that can assist you in troubleshooting. They are trace configurations and troubleshooting traces. You can configure three types of traces: SDI, SDL, and Log4j. The trace configuration option you pick has a big impact on the information you will receive and how much time you will probably spend during problem resolution. You can see what type of traces you can configure by checking out Figure 3-2. The files can be viewed in RTMT:

■ Configuration: Configure traces.

■ Troubleshooting Trace Settings: Set up predefined troubleshooting traces.

You are probably wondering how to choose between a trace and troubleshooting trace. The following table lists the main difference between these two.


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Traces Troubleshooting Traces

Minimal information, could miss relevant information

Lots of information, more than is needed to troubleshoot a specific problem

Apply filters Turns on almost all trace optionsLess system impact Major system impact

Trace types include the following:

■ SDI

Runtime events for the related software

IP address

Time stamp

Device name

■ SDL

Call processing from Unified Communications Manager and Cisco CTI Manager Services.

Used by Cisco TAC engineers. Normal administrators provide this information to them only upon request.

■ Log4J

Used for Java applications

Note: Traces affect system performance, so use them only for troubleshooting purposes. If you want to practice traces, use a nonproduc-tion system. In addition, SDL traces can be more difficult to read than SDI. Therefore, you usually start with SDI traces to find logs related to CUCM issues.


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FigURE 3-2 Where and How You Set Up Traces

To configure a trace, follow these steps:

1. Select the server.2. Select the service group from the drop-down.3. Select the service.4. Check the trace boxes you want.5. Click Save.


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Troubleshooting Trace SettingsWhen you need or want to run predefined traces, this where to go. The information you get here is detailed. You must search through files to locate the specific information. The required steps to implement this are as follows:

1. Click Trace.2. Click Troubleshooting Traces Settings.3. Select the server.4. Choose the services you want.5. Click Save.

Key Points for Traces■ Viewed with RTMT.

■ Can have major impact on system.

■ Output detail levels are either detailed or not very detailed.

■ SDL is used by Cisco Engineers for debugging. You will not use them, but if you do, they walk you through the process.

Example 2Figure 3-3 provides a sample SDI trace of a Cisco 7970 in my home lab failing to register. The information given is DeviceTypeMismatch, the MAC address, IP address, and a brief sentence to help you narrow the cause.


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Example 3An SDI trace of keepalives being sent from IP phones to Unified Communications Manager, the trace in Figure 3-4 shows that the keepalive is being sent every 30 seconds.

FigURE 3-3 SampleSDi Trace from a Cisco 7970 IP Phone


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Traces can provide valuable information to help resolve most, if not all, issues you might encounter. Cisco seems to have put considerable thought into how and what you are shown. Given the previous examples, you can easily piece together what is happening. In the earlier examples, you are given keywords, IP addresses, MAC addresses, a short sentence, and time stamps.

Note: Again, do not use traces in a produc-tion environment unless there is a problem. They can adversely affect your systems. All traces in this Quick Reference were done in a home lab.

FigURE 3-4 Trace Shows Keepalives Between Cisco IP Phones and CUCM


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Serviceability Sections

Service ActivationIn this section, you activate or deactivate services, which are pretty easy to work with. The services relating to the Unified Communications Manager you want to work with can be started, stopped, or activated. All you need to do is check the box that contains the service and save. The Service Activation page displays the status of all services. If a service is malfunctioning, you can check this page to see whether service is activated.

This page has a related link in the upper-right corner to quickly transition between two or more control centers.

Control CenterThe Control Center is just what it sounds like: It controls the state of services and features, displays status, and shows uptime. The Control Center has two sections that provide a good amount of detail that is handy for problem resolution: Network Services and Feature Services.

Network ServicesYou can start, stop, or restart network services here. The information displayed is the status of the service; see what time the service was started, and how long the service has been running.

If you are having network issues, this is the section of serviceability you will be checking. Network services are required for the operation of the server, not just Unified Communications Manager.

To activate/deactivate a service, follow these steps:

1. Log in to Cisco Unified Serviceability.2. Select Control Center, Network Services.3. Scroll down to server you want to manage.4. Click the services you want to manipulate. You can select more than one.


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Section 3 Cisco Unified Serviceability5. Click Save.

Feature ServicesThe Control Center Feature Services section is where you select the features you want your server to run. The features can be activated or deactivated. Which features are activated depends on the role of the server. If you are doing problem resolution, make sure that you’re troubleshooting the correct server.

SNMPThe Simple Network Management Protocol (SNMP) supported in Unified Communications Manager are v1, v2, and v3. If it has to do with SNMP, you will locate or configure it under the SNMP menu.

Dialed Number AnalyzerDialed Number Analyzer (DNA) is an awesome tool for checking to see whether your dial plan is valid and whether it is routing the way you want. For DNA to appear on the menu bar in Serviceability, you must first activate the feature in the Control Center. DNA provides a good amount of detail about the various components of the route plan and dial plan. All the DNA output pages look the same and provide the same information. They just test the route from different starting points. The one thing that differs is when selecting an IP phone you can specify which line on the phone. This is useful when you are dealing with device and line Calling Search Space (CSS). Figure 3-5 shows the options available when using Dialed Number Analyzer, while Figure 3-6 is an output from Dialed Number Analyzer.

Example 4This is a DNA of directory number 2003 trying to call 714-545-1111:

Results SummaryCalling Party InformationCalling Party = 2003Partition =

Note: Network services can only be started, stopped, or restarted.


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Device CSS =Line CSS =AAR Group Name =AAR CSS =Dialed Digits = 7145451111Match Result = RouteThisPatternMatched Pattern InformationPattern = 714545XXXXPartition =Time Schedule =Called Party Number = 7145451111Time Zone =End Device = SoCal_RLCall Classification = OffNetInterDigit Timeout = NODevice Override = DisabledOutside Dial Tone = NOCall FlowTranslationPattern :Pattern= = 7145451111 = 2003

= NO = 2003

Route Pattern :Pattern= 714545XXXXPositional Match List = 7145451111DialPlan =Route FilterRequire Forced Authorization Code = NoAuthorization Level = 0Require Client Matter Code = NoCall Classification =PreTransform Calling Party Number = 2003PreTransform Called Party Number = 7145451111Calling Party TransformationsExternal Phone Number Mask = NO


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Calling Party Mask =Prefix =CallingLineId Presentation = DefaultCallingName Presentation = DefaultCalling Party Number = 2003ConnectedParty Transformations = Default = DefaultCalled Party Transformations

= None = 7145451111Route List :Route List Name= SoCal_RL :SoCal_RG = 2003 = 7145451111

= Default = 2003

= 7145451111 := H323Gateway = 192.168.1.55 = 0 = UnKnown = Disabled = Disabled = OffNet = None

= 2003 = Originator = Default = = 2003 Alternate Matches


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The preceding example was created with a basic route established with all the default options. In a real-world environment, the CSS, partition, and most other field values are populated.

FigURE 3-5 DNA Main Page


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Having problems with your calling routes? DNA in Serviceability and the route plan report in Unified Communication Manager administration are the first places to start looking to resolve problems. These tools are crucial for an accurate route plan. Figures 3-5 through 3-8 can be examined to check routing behavior.

FigURE 3-6 Dialed Number Analyzer Results Page


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FigURE 3-7 Detailed Glimpse of a Dialed Number Analyzer


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The key troubleshooting features of DNA are as follows:

■ Calling number

■ Called number

■ Match result (Is the pattern routed or blocked?)

■ Route pattern selected

■ Route list used

FigURE 3-8 Another Look at Dialed Number Analyzer


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■ Call classification

■ Ability to test individual lines

■ Test route plan before going live

■ CSS and partition

Serviceability Reports ArchiveDaily reports are generated and accessible from this page. This is not really one of the better tools for troubleshooting, but it can provide information for extended periods of time.

The key troubleshooting points about Unified Serviceability are as follows:

■ Controls what network services and features are running.

■ Provides status.

■ Displays uptime.

■ Network services can only be started, stopped, or restarted.

■ Features services can be activated or deactivated.

■ Alarms are configured here.

■ Traces are configured here.

■ SNMP configuration v1, v2, and v3.

■ Trace files that are viewed with RTMT.

■ Database replication reports are available.


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Additional Resources■ World Wide Web: A useful place to help you figure out, resolve, and gather more information is online on the web. Like

most companies, Cisco has created a solid website to provide key information and resources to customers and anyone wanting to learn about Cisco products. If you are reading this guide, you have probably been to Cisco.com a few hundred times.

■ Cisco Search Engine: Search the entire Cisco indexed website.

■ Cisco TechTips: Contains Cisco TAC’s most common issues. Chances are if you are having problems, someone else has also experienced it. The issue may have been resolved and discussed in detail.

■ Output Interpreter: Here, you can analyze “some” show commands to assist in your troubleshooting.

■ The Cisco Support Community: Here, you can post questions, answer questions, and network with your technical peers.

■ Cisco Technical Assistance Center (TAC): When all else fails, check the Cisco TAC or contact them.


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Section 4

Command-Line Interface

The Cisco Unified Communication Manager comes with command-line interface (CLI) for fundamental functionality, such as maintenance, disaster recovery, and basic troubleshooting. The CLI is accessed by a connected terminal or through a Secure Shell (SSH) connection. Just like Cisco IOS, there are many commands; the entire command reference, however, is beyond the scope of this Quick Reference.

The main CLI command groups are as follows:

■ show: This command, used in conjunction with other options, displays some type of information, whether it is network, system, performance, or so on.

■ utils: This command is used to monitor various aspects of the server and Unified Communications Manager.

■ set: This command, with its options, is used to change values for the system. You can change thing such as IPv4 addresses, IPv6 addresses, password options, and so on.

■ file: This command, along with its options, manipulates files, by finding, deleting, and integrity checking.

As with Cisco IOS, detailed help is available with the help command.

When using a command, you can use ? to show more options if you are stuck or not familiar with the commands.

Cisco iOS Troubleshooting ToolsThe voice networks of today have many different components that interact to handle calls, provide features, and interact with other devices. To maintain and resolve issues you need to have an idea of what tools are available.


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Section 4 Command-Line Interface

Cisco iOS gatewaysGateways provide transitions between different networks. An example of this is Voice over IP (VoIP) to time-division multiplexing (TDM). Gateways are configured to communicate with the Unified Communication Manager, gatekeepers, or other gateways. Cisco gateways contain either analog ports or digital ports. Many gateways contain dial peers, which act like route statements for voice traffic. Your ability to analyze, troubleshoot, and resolve dial peer issues quickly using tools and debugs will be important throughout your career.

Dial PeersDial peers are an important component for VoIP on an IOS gateway. They determine the route or path to be taken and how the gateway handles the call.

It is important to know that plain old telephone system (POTS) dial peers show either UP or DOWN for their MIN and OPER or operational status fields. If you suspect dial peers are a source of your errors, check the MIN and OPER value. These values should be up/up during an active voice call. Here, the POTS dial peer and VoIP dial peer 61102 was shut down to demonstrate this principal:

FigURE 4-1 This figure shows that all Dial-peers are operational


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Section 4 Command-Line Interface

Cisco iOS debug CommandWhen using debug commands on a Cisco IOS gateway, you can drastically impact the performance and even cause the device to crash. This is not a good scenario unless you are on a nonproduction device. Here are a few common debug commands. I highly suggest that you search and know exactly which debug commands to use in a production environment:

debug voip dialpeer shows dial peer information.debugh225 asn1 | events shows the contents of messages pertaining to H225 RAS.debug h225 q931 shows q931 content in h245.debug isdn q931 shows isdn l3 or q931 signaling on a PRI.debug voice ccapi inout shows all information and debugs that pertain to the call control API.debug ccsip messages shows debug information related to Cisco Unified Border Element (CUBE) or Session Initiation Protocol (SIP) gateways.

Note: no debug all turns off all debugging.

FigURE 4-2 This figure shows two Dial-peers that have been shutdown


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Section 5

Troubleshooting Endpoint Registration

The Unified Communication Manager is responsible for registration and call routing for end devices, such as IP phones. In this section, we discuss how to troubleshoot registration issues with Skinny Client Control Protocol (SCCP) and SIP endpoints.

SCCP/SiPUnderstanding the boot process of the IP phone is invaluable to the resolution of any registration issues. By knowing the steps an IP phone takes during registration, you can quickly isolate and resolve many issues:

1. The phone receives power.2. The stored phone image is loaded.3. The voice VLAN is configured via Cisco Discovery Protocol or by hard setting on the phone.4. The IP address is assigned, and option 150 is learned for the TFTP address.5. Request configuration and profile from TFTP.6. Register with the primary Unified Communication Manager.

Because we are discussing registration, we start at Step 4, because Steps 1 through 3 are automatic and do not deal with registration. Your phone must have an IP address, be able to pull its configuration file, and finally register with the Unified Communications Manager.


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Section 5 Troubleshooting Endpoint Registration

iP: DHCP or StaticAs many of you know, IP addresses can be assigned either dynamically or statically. When the address is statically configured, the issues are misconfiguration or human error. In addition, this approach makes moves, adds, and changes difficult in many organizations because an administrator will need to physically reconfigure the phone. Dynamically configured IP addresses have a few more options that can cause failure.

Dynamic Host Configuration Protocol (DHCP) configuration errors are easy to recognize and resolve. Assuming your network is up and running, try ideas addressing the following troubleshooting assumptions:

■ DHCP scope has issued all addresses.

■ The IP phone and DHCP server are not on the same subnet, and DHCP relay is not enabled on the router, switch, or L3 interface.

■ The DHCP server scope is not defined for the subnet of the IP phone.

After you receive an IP address, the next issue you face is configuration files. The configuration files are sent from the TFP server. Phones not getting their configuration files should isolate the issue at the TFTP server. Check the following:

■ TFTP server is running, activated, and not hung.

■ Option 150 is configured for DHCP scope.

■ Option 150 is pointing to correct address in Unified Communication Manager (usually the Publisher).

■ Option 150 is configured and correct on the router.

Cisco iP Phones to the RescueCisco IP phones contain messages that are usually ignored, but they prove useful when problems arise. These little gems tell you what the IP phone is trying to do and what others (TFTP, VLAN, Unified Communications Manager) are saying back. You can view phone-related messages by pressing the Settings key on an IP phone.


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Section 5 Troubleshooting Endpoint Registration

The Cisco Unified Communication Manager can also help resolve some registration issues. The UCM has a few ways to register phones, either via auto-registration and manually. There are times when the phone should be working and, for some reason, you just cannot get it to work. One common error message is

Registration Rejected: Error DBConfig Error Message

Typically, a Registration Rejected: Error DBConfig message indicates an issue with unassigned DNs:

1. Delete the DN in the Directory Number section.2. Delete the phone in the Device section.3. Manually re-add the phone or try to let it autoregister. This should solve the remaining issues.

H.323 EndpointsH.323 endpoints do not show registered status in Unified Communication Manager. You must manually configure the H.323 endpoints. The dial plan and dial peers, along with configuration elements for H.323, provide an autonomous system that can process calls in the event the CUCM servers are down.

gateway RegistrationsGateways need to have communication paths that remain open with Unified Communication Manager. The type of gateway used determines the process and commands needed to register.


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Section 6

MGCP

Media Gateway Control Protocol (MGCP) must be configured on both the gateway and the Unified Communication Manager. Here are the steps needed to register an MGCP gateway to a CUCM server:

1. Configure the IP address on the interface and enable IP routing on the gateway.2. Give a unique name to the gateway:

hostname ATL1-mgcp

3. Start the MGCP daemon:mgcp

4. Configure the IP address or DNS for CUCM:mgcp call-agent 10.10.10.1 (This is the IP address of call control agent (CUCM).)

5. Enable CUCM support in MGCP:ccm-manager mgcp

6. Enable the gateway to download configuration files:ccm-manager configccm-manger config server (ip address)


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Section 6 MGCP

Unified Communication Manager MgCP ConfigurationThe configuration and troubleshooting of MGCP is fairly easy on the UCM. The first step is to configure the gateway and the ports on the gateway: Here are the steps needed to configure MGCP, Figure 6-1 shows the actual page on CUCM.

1. Choose Device, Gateway.2. Click Add New.3. Select a gateway type.4. Configure the gateway and module information.5. Click Save.


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Section 6 MGCP

When you configure the gateway and communications are established, it shows as registered on the gateway page.

FigURE 6-1 Example of MGCP Gateway Configuration


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Section 6 MGCP

Troubleshooting MgCP■ Make sure the IOS gateway was manually configured for UCM.

■ The configuration of gateway modules on UCM is inaccurate.

■ If you are pulling files and configuration from UCM, check UCM and make sure that configuration under the gateway is accurate.

■ Check whether Cisco IOS and Cisco Unified Communication Manager software is compatible.

■ show ccm-manager shows the status of the gateway.

■ In UCM, under Gateway, a status of registered or unregistered shows.

The process of MGCP working with UCM involves a few simple steps configured on the MGCP gateway or UCM. The chance that problems are configuration related is around 90+%. The remaining issues will be either network or some unforeseen issues.

Monitoring MgCPYou can use the following commands to monitor the status of MGCP components:

show ccm-manger shows the primary and backup server, keepalive interval, switchback mode, and traffic information.show mgcp endpoints shows the state of endpoints.show mgcp connections shows the status of active calls on registered endpoints.show mgcp statistics shows stats for sent and received MGCP messages. The key fields here are the failed sections.show mgcp srtp detail is used to get more detail on Secure Real-time Transport Protocol (SRTP).show mgcp srtp summary provides an overview of SRTP.


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Section 6 MGCP

MgCP debug CommandsThere are a few debug commands to isolate issues. As mentioned previously, be careful when using debug commands in a production environment:

debug mgcp packets provides information about MGCP packets and devices.debug mgcp state displays state changes on MGCP endpoints.no debug all turns off all debugging.


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Section 7

H.323 and SIP Gateways

The main thing to remember about H.323 and SIP gateways is that they do not register with Cisco Unified Call Manager. All call routing must be configured locally on the gateway. Therefore, you must make sure the configuration on both ends is correct. For problem resolution, the issues are either network or configuration related.

H.323 gatewaysWhen checking the gateway using IOS, make sure that the required configuration is on the gateway and that the gateways have connectivity to the needed devices:

interface FastEthernet0/0 ip address 67.43.157.125 255.255.255.128 ip route-cache same-interface duplex auto speed auto h323-gateway voip interface --This command is mandatoryh323-gateway voip bind srcaddr 67.43.157.125 --This command is mandatory

The following commands are used if you have gatekeepers:

h323-gateway voip id tpgk1 ipaddr 67.43.157.124 1719 h323-gateway voip h323-id HQ

When working with IOS gateways, make sure that you have the proper gateway configuration in place, as follows:


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Section 7 H.323 and SIP Gateways

1. Specify the VoIP interface:h323-gateway voip interface

2. Specify IP to be used for communication to UCM:h323-gateway voip bind srcaddr 67.43.157.125

The following two commands are used if your network contains gatekeepers.3. Specify the gatekeeper IP and ID:

h323-gateway voip id tpgk1 ipaddr 167.143.155.124 1719

4. Configure H.323 ID for the gateway:h323-gateway voip h323-id HQ

The next command to use is the show gateway command, which displays information on the status of the gateway and which gatekeepers it is registered with. Here is an example of a gateway not configured to register with a gatekeeper (Step 2):

HQ# sh gatewayH.323 ITU-T Version: 4.0 H323 Stack Version: 0.1

H.323 service is up Gateway HQ is not registered to any gatekeeper

Alias list (CLI configured) H323-ID HQAlias list (last RCF) is empty

H323 resource thresholding is Disabled

When you are dealing with an H.323 gateway working with a Cisco Unified Communication Manager, enter the IP address of the Communications Manager in the session target of one of your dial peers.


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SiPThe command show sip-ua calls the shows active user agent client (UAC) and the user agent server (UAS) information for active calls:

HQ# show sip-ua callsSIP UAC CALL INFO

Number of SIP User Agent Client(UAC) calls: 0SIP UAS CALL INFO

Number of SIP User Agent Server(UAS) calls: 0

HQ#

The show sip service command shows SIP service information.

The debug ccsip messages command shows SIP messages exchanged between UAC and the access server:

HQ# sh sip serviceSIP Service is up

The show sip-ua statistics command shows traffic, response, and retry information on SIP:

HQ# sh sip-ua statisSIP Response Statistics (Inbound/Outbound) Informational: Trying 0/0, Ringing 0/0, Forwarded 0/0, Queued 0/0, SessionProgress 0/0 Success: OkInvite 0/0, OkBye 0/0, OkCancel 0/0, OkOptions 0/0, OkPrack 0/0, OkPreconditionMet 0/0, OkSubscribe 0/0, OkNotify 0/0, OkInfo 0/0, 202Accepted 0/0


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Redirection (Inbound only except for MovedTemp(Inbound/Outbound)) : MultipleChoice 0, MovedPermanently 0, MovedTemporarily 0/0, UseProxy 0, AlternateService 0 Client Error: BadRequest 0/0, Unauthorized 0/0, PaymentRequired 0/0, Forbidden 0/0, NotFound 0/0, MethodNotAllowed 0/0, NotAcceptable 0/0, ProxyAuthReqd 0/0, ReqTimeout 0/0, Conflict 0/0, Gone 0/0, ReqEntityTooLarge 0/0, ReqURITooLarge 0/0, UnsupportedMediaType 0/0, BadExtension 0/0, TempNotAvailable 0/0, CallLegNonExistent 0/0, LoopDetected 0/0, TooManyHops 0/0, AddrIncomplete 0/0, Ambiguous 0/0, BusyHere 0/0, RequestCancel 0/0, NotAcceptableMedia 0/0, BadEvent 0/0, SETooSmall 0/0 Server Error: InternalError 0/0, NotImplemented 0/0, BadGateway 0/0, ServiceUnavail 0/0, GatewayTimeout 0/0, BadSipVer 0/0, PreCondFailure 0/0 Global Failure: BusyEverywhere 0/0, Decline 0/0, NotExistAnywhere 0/0, NotAcceptable 0/0 Miscellaneous counters: RedirectRspMappedToClientErr 0

SIP Total Traffic Statistics (Inbound/Outbound) Invite 0/0, Ack 0/0, Bye 0/0, Cancel 0/0, Options 0/0, Prack 0/0, Comet 0/0, Subscribe 0/0, Notify 0/0, Refer 0/0, Info 0/0


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Retry Statistics Invite 0, Bye 0, Cancel 0, Response 0, Prack 0, Comet 0, Reliable1xx 0, Notify 0 Info 0

SDP application statistics: Parses: 0, Builds 0 Invalid token order: 0, Invalid param: 0 Not SDP desc: 0, No resource: 0 Last time SIP Statistics were cleared: <never>


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Section 8

Troubleshooting Database Issues

The database of Unified Communications Manager is an IBM Informix database that holds all the configurations, settings, service parameters, and the dial pan. It is vital to the operation of your voice network that the database be functioning and available to send and receive. The publisher replicates its database (master) to all the subscriber nodes. There are two types of replications methods: one-to-many or mesh mode. Most replications use the one-to-many. When changes are made other than replication, however, such as new devices or resources being added, mesh mode is used to allow for optimal routing. In this section, we look at some of the common causes of replication issues and ways to resolve them.

Network ConnectivityWhen a problem occurs in your network where nodes cannot communicate, your nodes will have outdated information:

■ Check network connectivity.

■ If Domain Name System (DNS) is used, ensure that name resolution is working.

■ No more than 80 ms round-trip delay between nodes.

Oversubscribed HardwareThe CPU has no CPU cycles/power to perform replication. Another cause could be no memory to allocate:

■ Check performance monitors to verify and consider upgrading equipment.


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Section 8 Troubleshooting Database Issues

To check for database replication issues, you can look in RTMT, CLI, or Cisco Unified Reporting. These three areas show the cluster replication information, more specifically the Replication_State object indicator:

0 = Replication did not start.1 = Replicates create, but their count is incorrect.2 = Replication is good.3 = Replication in cluster is bad.4 = Replication setup failed.

admin:utils dbreplication status-------------------- utils dbreplication status --------------------

Status cannot be performed when replication is down on the publisher or on a cluster with a single active node or a core’s node, which aborts the replication status check operation.

Use the following command to see the output:

file view activelog cm/trace/dbl/sdi/ReplicationStatus.2011_01_30_23_30_51.outadmin:

The preceding command is a CLI example of the command utils dbreplication status, as you can see by the output database replication being turned off on this UCM. Figure 8-1 shows the RTMT example of the same database replication information.


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CLi Database Commandsutils dbreplication clusterreset resets replication on the entire cluster.utils dbreplication dropadmindb drops the Informix syscdr database on any server in the cluster. (Caution: Run this command only if the command utils dbreplication reset or utilsdbreplication clusterreset fails and replication cannot be restarted.)utils dbreplication dropadmindbforce is basically the same as dropadmindb. Run only if requested by Cisco TAC.utils dbreplication forcedatasyncsub forces a subscriber server to have its data restored from data on the publisher server.utils dbreplication quickaudit runs a quick database check on selected content on dynamic tables.

FigURE 8-1 RTMT Performance Counters


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utils dbreplication repair attempts to repair mismatched data.utils dbreplication repairreplicate repairs mismatched data between cluster nodes, and changes node data to match the pub-lisher.utils dbreplication repairtable is basically the same as repairreplicate.utils dbreplication reset resets database replication.utils dbreplication runtimestate monitors progress of the database replication process and provides replication state in the clus-ter.utils dbreplication setrepltimeout increases the default timeout for replication setup on large clusters.utils dbreplication status shows status of database replication after setup is complete.utils dbreplication stop stops the automatic setup of database replication and used to stop the ongoing setup of replication. Run on each node where replication needs to be stopped.

If you confirmed database replication issues, try to repair the database:

1. Run utils dbreplication repair in the CLI.2. Verify the status when complete.

After you try to repair the database, if you still receive a status of 0 or 4, or if the problem is not resolved after four hours, Cisco recommends that you reset the database, as follows:

1. Stop the replication between the publisher and problem subscriber.2. Reset the replication of the failed subscriber.3. Verify the issue was resolved. Check status.

To reset database replication, follow these steps:

1. On the subscriber, enter utils dbreplication stop in the CLI. (This is done on all status four subscribers.)2. On the publisher, enter utils dbreplication stop for all affected subscribers.3. On the publisher, enter utils dbreplication reset.4. Use the command utils dbplication runtimestate show SYNC COMPLETED.5. Verify with utils dbreplication status.


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To reset the cluster, follow these steps:

1. On all affected subscribers, enter utils dbreplication stop in the CLI.2. On the publisher, enter utils dbreplication stop.3. On the publisher, enter utils dbreplication clusterrest.4. Monitor /var/log/active/cm/trace/dbl/sdi for status.5. On the publisher, enter utils dbreplication reset all.

When the status shows 2, stop and restart all services on subscribers.


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Section 9

Troubleshooting Call Setup Issues

When troubleshooting calls, the type of deployment model used can help isolate what might be causing the issue. The art of troubleshooting any type of call setup is in knowing where the call is coming from, who is being called, and who has permission to call whom. When these things are known, you can start to determine where to begin.

We begin by looking at the common call setup issues:

■ Dead air

■ No ring back

■ Missing caller ID or no caller ID

■ Fast busy tone

■ Unexpected second dial tone

intrasite Call SetupSingle-site deployments are naturally easier to resolve because they have fewer pieces of the puzzle and less-complex dial plans. All devices are considered local to the Unified Communication Manager. This guide assumes you are familiar with the workings of how line and device CSS, partitions, and time-of-day (ToD) routing work. I do not detail how they operate except in relation to troubleshooting.


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Section 9 Troubleshooting Call Setup Issues

Some of the scenarios you might see in single-site deployment are fast busy, reorder tone, call forwarding problems, and one-way calling. Probable causes of such issues include the following:

■ Incorrect CSS

■ Incorrect partitions

■ Device not registered to UCM

■ Incorrect forwarding options

If you are experiencing call setup issues in a single-site deployment, check the following:

■ The device has permission to call the other device. Check CSS and partitions.

■ No translations pattern loops exist.

■ CTI ports are configured correctly.

■ Called device is registered with UCM.

■ Voice mail ports are configured correctly and not all busy.

■ If directory number is not registered and Call Forward Unregistered (CFUR) is not configured.

The key points to remember include the following:

■ By default, most fields in line or device configuration are set to default value.

■ Line CSSs takes precedence over device CSSs. Here’s an example:

Line 1 CCS, line 2 CSS, and then device 1 CSS is the order chosen.

■ Partitions are listed in highest-priority order.

■ Any device/port/trunk in the <none> partition is accessible from anywhere regardless of CSSs.

■ ToD: CSSs have access to the partitions with ToD configured only if the time and date match.


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Dialed Number Analyzer (DNA), as discussed earlier, can be a good tool to show which translation patterns are being used, whether translation patterns are changing digits, which CSS and partitions are being used, and whether ToD routing is taking place.

Figure 9-1 shows the CSS and partition of the device placing the call. What it does not show is that the call gets fast busy (because it has no permission to call any other phones, just to receive calls).

FigURE 9-1 Phone Configuration and Output for Dialed Number Analyzer


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The Cisco Unified Communication Manager is another resource to check issues. The Directory Numbers page in UCM should be checked. On this page, make certain that the DN is registered and valid. Figure 9-2 shows the Directory Numbers page.

FigURE 9-2 CUCM Directory Numbers Page


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intersite Call Setup issuesWhen troubleshooting multisite deployments, you have more pieces that can cause problems. A few areas where problems can happen are dial plans, class of service (CoS), the way UCM handles the dialed digits, translations, date/time restrictions, resources not available, and bandwidth. The dial plans and calling permissions (CoS) are more complex, which means they can be more difficult to locate.

Common problem areas include

■ CSS

■ Partitions

■ ToD routing

■ Translation patterns

■ MTP, coding, conference, termination resources

One-Way CallingOne-way calling is when user A at DN 2001 can call user B at DN 2002. When user B tries to call user A and the call fails, this is a CoS issue where user B does not have permission to call user A.

Call Forwarding issuesA user has her calls forwarded to voicemail or a different user when she is on the phone or away from her desk. She calls the help desk and says that she can receive calls fine but that when calls are sent to voicemail or another coworker the calls do not work. Well, we know that she can receive calls, but the forwarding of calls and voicemail fail. We can assume that because voicemails are basically just unanswered calls forwarded to voicemail, the culprit must be forwarding problems. All call routing targets have permissions assigned to them. The permissions can be default permissions or they could be administrator or even used defined. Is the target registered?


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FigURE 9-3 What Number Is the Dialed Number and How the System Handles the Call


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Trunks/CircuitsIntersite calls will most likely be using some type of trunk or circuit to transport your calls. The process of sending calls offsite could involve one more devices. Each of these devices has components that also need to be configured. As stated previously, all route targets have some type of CoS configured. Luckily for us, the same basic issues manifest themselves across multiple sites.

Another potential problem area with trunks and circuits is the issue of oversubscribed links. When WAN links get busy calls, they can suffer choppy voice, disconnected calls, or even calls setup failure. You must manage your WAN bandwidth. A few devices and components can relieve some of these problems.

Common trunk/circuit/WAN issues include the following:

■ Oversubscribed circuits

■ Codec mismatch

■ CoS

■ Service provider problems

■ Hardware resources (MTP, conference bridges)

RegionsUCM has a section in the administration pages called Regions. A region allows you to configure the codec used and the maximum video bandwidth (in kilobits per second) for calls between sites.

gatekeepersGatekeepers provide call admission control (CAC), usually for trunks and H.323 devices. When used, gatekeepers can provide address translation, admission control, zone management, and bandwidth control.

gatekeeper MessagesAll gatekeeper messages have the same basic function: a request (RQ), confirmation (CF), or reject (RJ). A request will be made, and then that request is either confirmed or rejected. If you keep this in mind, you should be able to decipher what you see in output.


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H.225 RAS messages are as follows:

■ Discovery (GRQ, GCF, GRJ)

■ Admission (ARQ, ACF, ARJ)

■ Registration (RRQ, RCF, RRJ)

■ Unregistration (URQ, UCF, URJ)

■ Location Request (LRQ, LCF, LRJ)

■ Disengage (DRQ, DCF, DRJ)

■ Resource Availability (RAI, RAC)

■ Request in Progress (RIP)

■ Status Queries (IEG, IRR, IACK, INACK)

■ ACK = Acknowledge

gatekeeper iOS Commandsshow gatekeeper status displays gatekeeper and zone status.show gatekeeper zone status shows zone information.show gatekeeper endpoints shows the registered endpoints.show gatekeeper calls shows the status of each outgoing call the gatekeeper knows about.show gatekeeper zone prefix shows the zone prefix table.show gatekeeper gw-type-prefix shows the configured prefix technology table.

Here is a basic gatekeeper configuration example:


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gatekeeper zone local gk1 telepacket.com invia gk1 outvia gk1 zone remote gkm gkm.voip.com 102.70.109.62 1719 invia gkm outvia gkm zone cluster local gk_cluster gk1 element gk2 77.143.108.121 1719 no shutdown

zone local defines the local zone and IP address.zone remote defines a remote zone and IP addresszone cluster defines the local group of gatekeepers.

When troubleshooting gatekeeper issues, there should be a reason for rejection. This will assist in problem resolution.

Example 5User A is trying to call client R in NYC. You, as the administrator, must take into account all the CSSs, partitions, and ToD for the line and phone. In addition, we have added the route list, route groups, route patterns, and translation patterns. When those items are working, you must decide whether you need to add/subtract digits before sending out the call. Everything works fine if we are handing off the call to a service provider.

Now flip this around and consider if you, the administrator, must also manage the destination site. All the same components need to be configured for the call to complete. You must be able to follow (trace) the call from start to finish.

Note: The invia and outvia commands are the addi-tional gatekeepers used to enter or leave a zone.


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Section 10

Tracing Calls

You will sometimes need to see the entire picture of what is happening to fix problems. UCM has a tool that allows just that: trace. Consider this example: Taylor, my oldest child, is at DN 2005. She is calling her sister Brittney at DN 2003. Taylor, being the oldest, can call anyone she wants. Nathan is the youngest and has no rights to call anyone. (Nathan is not calling anyone in this example.)

22:14:09.438 |Digit analysis: match(pi=”2”,fqcn=”4042172005”, cn=”2005”, plv=”5”,pss=”TaylorPT:BrittneyPT:NathanPT”, TodFilteredPss=”TaylorPT:BrittneyPT:NathanPT”,dd=”200”,dac=”0”)|1,100,49,1.5271^192.168.2.201^SEP001FCA34900822:14:09.438 |Digit analysis: potentialMatches=PotentialMatchesExist|1,100,49,1.5271^192.168.2.201^SEP001FCA34900822:14:09.924 |StationInit: (0000220) KeypadButton kpButton=3.|1,100,49,1.5273^192.168.2.201^SEP001FCA34900822:14:09.928 |Digit Analysis: star_DaReq: daReq.partitionSearchSpace(f170d664-4c96-b893-8a75-79b3d78f112f),filteredPartitionSearchSpaceString(TaylorPT:BrittneyPT:NathanPT),partitionSearchSpaceString(TaylorPT:BrittneyPT:NathanPT)|1,100,49,1.5273^192.168.2.201^SEP001FCA34900822:14:09.928 |Digit Analysis: star_DaReq: Matching Legacy Numeric,digits=2003|1,100,49,1.5273^192.168.2.201^SEP001FCA34900822:14:09.930 |Digit analysis: match(pi=”2”, fqcn=”4042172005”, cn=”2005”,plv=”5”,pss=”TaylorPT:BrittneyPT:NathanPT”, TodFilteredPss=”TaylorPT:BrittneyPT:NathanPT”, dd=”2003”,dac=”0”)|

filteredPartitionSearchSpaceString(TaylorPT:BrittneyPT:NathanPT) is the partitions Taylor has permissions to and are in the final CoS descision.Matching Legacy Numeric,digits=2003 shows that digit matching is done and a number has been selected.dd=”200” shows the first three digits dialed.dd=”2003” shows all four digits dialed.


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Section 10 Tracing Calls

This example shows just a few areas of a trace that you should look at to help resolve complex call routing problems.

Example 6This is a part of Brittney’s trace showing her calling permissions. Notice that TaylorPT is not there; Brittney does not have permission to call her sister:

22:14:53.791 |Digit Analysis: star_DaReq: daReq.partitionSearchSpace(197277b3-81e7-f802-0412-369af8f41bad),filteredPartitionSearchSpaceString(BrittneyPT:NathanPT),partitionSearchSpaceString(BrittneyPT:NathanPT)|1,100,49,1.5317^

The key points about trace include the following:

■ Use the search feature to find information quick.

■ Search for the IP of the device.

■ Search for filteredPartitionSearchSpaceString.

The three tools you use to locate the issues are DNA, UCM administrative web pages (Directory Numbers, Device, CSS, and Partition sections), or you can pull the trace files using RTMT. DNA and UCM admin pages should provide more than enough information to quickly resolve most single-site issues, along with some multisite ones. Traces should be used in complex situations so that they can more quickly provide the “big picture.”


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Section 11

Troubleshooting Off-Net Call Setup

When dealing with off-net call setup issues, all the previously stated components and issues are still present. Now, we just add a few more components and configuration sections. Naturally, when packets leave your control, your VoIP call is at the mercy of others. The same is true for off-net calls. The administrator of your company’s network has little, if any, control of packets after they leave the comfort of the company-controlled network.

Common off-net issues include the following:

■ Gateway configuration errors

■ CoS configuration errors

■ Service provider issues

■ Codec settings on the Region page

■ CAC location settings

■ QoS setting for WAN links

■ Wrong dial peer session target

After reading the common issues for off-net calls, you may think, “Hey, I know where to start looking to resolve this.” I agree. At this point, you should have a solid understanding of all the tools and resources to resolve most issues. The old saying “more money, more problems” comes to mind. Yes, we have a few more areas to look into, but the areas are the same as always. Granted, a few new ones have popped up. The old problems are the same, and we already know how to solve those (or at least where to start solving them). This makes coping with the new problems easier because we can devote more time to them.


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Section 12

Troubleshooting Voice Quality Issues

Voice quality is defined as the quality of a call. No, I do not mean if the other person is boring you; I mean how clear the sound is on the call. Does it take a few seconds for the other party to hear what was said?

Many things can affect the quality of the call:

■ One-way audio

■ Echo

■ Choppy voice

■ Robotic-sounding voices

The list could go on and on. Here, we discuss the more common issues and how to resolve them.

One-Way AudioThis is when caller A can hear caller B, but caller B cannot hear caller A.

The usual culprit for one-way audio relates to IP routing. Remember, not all packets take the same path to the destination. Packets A, D, and E might go across an OC-3 link that has no compression or any type of security functionality, whereas packets B and C go over a T1 line used by everyone. The T1 line uses header compression, access lists, and firewalls on both ends.


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Section 12 Troubleshooting Voice Quality Issues

Common one-way voice issues include the following:

■ IP reachability: Not all packets take same path. (In this case, check network connectivity for the affected end. Use an extended ping on the router.)

■ H.323 binding: When using H.323 and you have multiple active IP interfaces, your H.323 signaling and RTP streams may be sourced from multiple IP addresses. This can cause one-way audio. The solution is to use the following command:h323-gateway voip bind srcaddr ip-address

The command should be used on a loopback address, because loopbacks are the most stable interfaces.

■ Cisco IOS NAT version: The Network Address Translation (NAT) may cause one-way audio when using SCCP or H.323 on versions earlier than 12.1(5)T. The solution is to upgrade the IOS.

■ Firewalls: A firewall could be blocking RTP traffic coming in, but allowing it out. The solution is to check firewall rules.

■ VPN: Endpoints do not use the correct IP address. This solution is this: The VPN endpoint address should not use the adapter’s IP address.

Dead AirDead air is when your call connects and neither party hears any sound. This means that no RTP packets are being received. The call setup is not the same as the RTP conversation. Likely causes include the following:

■ Access list preventing RTP traffic: Checking the access list should be straightforward.

■ Codec mismatch: Check whether transcoding resources are needed; also try codec negotiations.

■ Firewall: Check firewall settings.


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Dropped CallsThis is one of the hardest things to troubleshoot, but also one of the easiest. The reason I say this is if your users are experiencing dropped calls at a certain time each day, you can reason that it’s the busy hour with too many calls to handle. What if you have a user who randomly drops calls? This is harder to resolve.

Possible causes of dropped calls include the following:

■ System or phone software failure: Something caused instability. Try resetting the phone or device.

■ Network connectivity: A network event (down circuit, incorrect routing information, and so on) occurred and impacted the RTP path. Check network connectivity.

■ Cell phone: One party to the call was on a cell phone and the call was dropped. Ask the user whether the other party was on a cell phone.

■ Carriers: The carrier experienced some sort of failure on their end of the network.

■ Codecs: Calls can also be dropped when their codecs are mismatched. The endpoints do not understand how to communicate with each other, so the call never takes place.

To troubleshoot dropped calls, do the following:

■ Check logs for events resets or restarts.

■ Check with the carrier to see whether it’s having issues on its end.

■ CDR for PSTN calls to see which end is disconnecting.

Second Dial TonesThis is fairly easy: Unified Communications Manager route patterns have a Provide Outside Dial Tone check box. Check your route patterns.


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EchoEveryone just loves it when echo occurs. No, seriously, echoes can be annoying and ruin phone conversations. There are two types of echo: electrical and acoustic. Electrical echoes result exclusively from analog circuits. Acoustic echo is generated by acoustic energy that enters the mouthpiece or microphone.

To isolate and deal with echo, you can do the following:

■ Identify the tail circuit causing the echo. The echo is caused on the tail circuit opposite of the party hearing the echo.

■ Check speakerphones and headsets, because hey are probably the cause. Replace them, if possible.

■ Check the remote gateway to see whether echo canceling is enabled. You can use the following IOS commands to do so:echo-cancel enabled on voice portsecho-cancel coverage on voice ports

Remember to issue shutdown and then no shutdown.

■ Change the signal coming into and leaving the gateway with the following IOS commands:input gain valueoutput attenuation value

Remember to issue shutdown and then no shutdown.


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Section 13

Cisco Service Advertisement Framework

Cisco Service Advertisement Framework (SAF) is reliable framework that allows applications to locate and get connectivity information about networked resources on a network. Applications can advertise (publish) or request (subscribe to) services on the network. SAF provides the IP address, configuration information, ports, and services that are present within the network.

Cisco Unified Communications Manager uses Call Control Discovery (CCD) to dynamically set up intercall agent information. CCD lets the UCM advertise DN patterns, features, and the UCM itself. This allows Unified Communication Managers to use advertised information to dynamically change their routing behavior and decisions.

The two components of SAF are

■ SAF forwarders: Receives services that SAF clients advertise and then distributes this information throughout the network. The forwarders make service information available for clients to use.

■ SAF clients: These entities advertise (publish) their services or request (subscribe to) services. SAF clients must register with SAF forwarders before they are allowed to participate in the SAF process. Clients send a register message to the forwarders, and a relationship is established. Each client has a unique ID to distinguish itself on the network.

There are two types of clients: internal and external.

All SAF transaction contains a request message followed by a response message. Message types are response or request. Response types are success or error.


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Section 13 Cisco Service Advertisement Framework

SAF methods are as follows:

■ Register, Unregister, RegisterRevoke

■ Publish/Notify, Unpublish/Withdraw, PublishRevoke

■ Subscribe, Unsubscribe

Service iDCCD for Unified Communications uses the following: 101:2:x.x.x.x

■ Service ID 101= Cisco Unified Communications

■ Subservice ID 2= CCD

■ Instance ID x.x.x.x = ID of UCM cluster primary key ID or IOS device

SAF uses Extended Interior Gateway Routing Protocol (EIGRP) to distribute service information on the network. EIGRP routing do not need to be enabled/running.

To configure EIGRP with the SAF client, use the following commands:

router eigrp saf enables the EIGRP virtual instance.service-family ipv4 autonomous-system 20 enables the SAF service family for the autonomous system.topology base enables the service family interface topology configuration.external-client cucm_pub configures the SAF external client; this must match in UCM.service-family external-client listen ipv4 5050 configures the external SAF client on port 5050.

external-client cucm_pubusername brionpassword voice123

The username and password configured on the router must match the values configured in the SAF security profile.

You must make sure IP address, port, and client name match the configuration on the gateway/forwarder. Figure 13-1 show the required information to configure a SAF security profile.


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Troubleshooting SAF RegistrationsRegistration is a big part of troubleshooting. For SAF components to work, the client must be registered with a forwarder. You can check registration status in two places: RTMT and via IOS.

RTMT’s SAF forwarders under the Reports section display which forwarders the UCM is registered with. Figure 13-2 shows what information can be obtained from SAF Forwarders in RTMT. As you can see, no SAF forwarders are configured, but you can still see the sections that would be populated.

FigURE 13-1 SAF Security Profile Configuration Page on the SAF Forwarder Configuration Page


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Here are the IOS commands:

show eigrp service-family ipv4 clients shows the registered clients for the forwarder.show eigrp service-family ipv4 clients shows registered clients for the forwarder in detail.

Key points include the following:

■ Configurations must be identical on IOS and the UCM.

■ RTMT and IOS show commands are used to provide facts.

Note: I am not currently running any SAF in my network, so the fields shown in Figure 13-2 are not populated.

FigURE 13-2 Example of RTMT SAF Forwarders


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SAF Client StatisticsThis is a decent tool for troubleshooting SAF. RTMT Performance provides an interface to view successful and failed connections. The command show eigrp service-family ipv4 traffic provides more information about what SAF traffic is doing.

Troubleshooting SAF AdjacenciesWhen SAF nodes cannot discover each other, they cannot form adjacencies. Possible reasons include the following:

■ SAF not enabled on routers. Solution: Enable SAF.

■ Firewalls are blocking hello packets. Solution: Check the firewall configuration.

■ Inaccurate configurations. Solution: Check your configurations line by line.

■ Network connectivity issues. Solution: Use ping and traceroute on the affected device.

Here are the IOS show commands:

show eigrp service-family iv4 neighbors shows neighbors and the interface they are neighbored on.show eigrp service-family ipv4 interfaces shows the interfaces using SAF.show eigrp service-family iv4 neighbors detail shows neighbors and the interface they are neighbored on in more detail.

Troubleshooting SAF External ClientsSAF forwarders need to be registered with external clients. When no communications occur, SAF cannot do its job.

Possible causes include the following:


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■ Incorrect external client SAF trunk configuration.

■ Trunk not SAF enabled.

■ Trunk not associated with CCD.

■ The activated feature’s check box is unmarked.

■ Credentials are incorrect.

■ The IP address or port is incorrect.

Here are some ways to resolve these issues:

■ Recheck that configurations and credentials match on both ends.

■ Check trunk configurations.

■ Enable trace, select Enable Forward and Miscellaneous Trace.

Troubleshooting SAF information ExchangeWhen SAF forwarders are not exchanging call control information, you need to check network and quality of service (QoS) sections.


■ Not becoming adjacent.

■ Split horizon is in action.

■ SAF bandwidth percent is set too low.

■ Authentication is present.

■ Network connectivity issues are present.

■ Max hop count is to low.


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Solutions include the following:

■ Check the interface bandwidth. Use the command bandwidth interface to change the value.

■ Verify whether authentication is used. Use the command show eigrp service-family ipv4 interface detail (serial 1/1).

■ Check hop count. Use the command show eigrp protocols to view the maximum hop count value.

■ Check topology. Use the command show eigrp service-family ipv4 topology.

■ Check whether network issues are present.

Troubleshooting Service Advertisement Call Control Discovery issuesKey points include the following:

■ Dynamically learned routes are not stored in the UCM database; they are stored in memory.

■ All SAF components are working.

■ SAF uses request and response.

Verifying Learned RoutesWhen troubleshooting learned routes, you need to use RTMT and IOS commands.

RTMT displays the learned patterns, the status of the patterns, and the PSTN failover pattern.

The show voice saf dndb all command shows all the learned routes on the IOS device; information includes the total number of routes, max allowed routes, and error statistics.


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Patterns Learning ProcessWhen patterns are not being learned, you have to start from the bottom up. The bottom, in this case, is that there could be the familiar network connectivity issue. Other possible causes include routes being blocked and perhaps that the maximum number of routes has been reached.


■ Unified Communication Manager and SAF forwarder are not in functional state. Solution: Verify that the SAF forwarder is registered with UCM.

■ CCD requesting the service is configured incorrectly. Solution: Check SAF forwarder values on UCM.

■ SAF not distributing call control information. Solution: Check and verify EIGRP state.

■ Pattern blocking on UCM is keeping routes from being valid. Solution: Check route patterns.

■ Max routes have been learned. Solution: Increase max route value.

Verifying Advertising and Requesting ServicesVerification of SAF services can be accomplished from the Call Routing, Call Discovery section within UCM. When checking, check the local and remote sides.

Tracing CCDWhen tracing CCD issues within Unified Communications Manager, you really just need to know where to turn on tracing and what field to trace.


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The key points are as follows:

■ Enable all phone devices.

■ Enable all gateway traces.

■ Enable Miscellaneous Trace.

■ Enable Forward and Miscellaneous Trace.

■ Use the Search function when dealing with the output.

Learned but UnreachableYour routes are showing up, but users cannot use the routes. Possible causes include the following:

■ Loss of connectivity. Solution: Restore connectivity.

■ Learned routes might not be included in the CSS of the phone. Solution: Add routes to the CSS of the phones.


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Section 14

Troubleshooting Device Mobility Issues

Users roaming away from their home office will experience issues. The issues they face will generally be the same ones that a user who does not roam also faces. However, mobility does risk a few more issues.

Common device mobility issues include the following:

■ CoS.

■ Codec and CAC.

■ Device Mobility is turned off.

■ SRST is pointing to the wrong IP address.

■ DHCP and IP addresses are misconfigured in Locations.

■ Call routing and AAR.

■ Media resources.

CoSClass of service (CoS) gives many administrators a headache. A typical scenario for CoS is a user has calling permissions that he should not have, while at the same time, he is missing permissions he should have.


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Section 14 Troubleshooting Device Mobility Issues

You can resolve this issue as follows:

■ Be very well versed in CSS.

■ Check the final CoS with a trace.

■ Check partition and CSS configurations.

■ The device pool configuration applied by the roaming site.

■ Make sure that your Device Mobility design is accurate.

Codec, CAC, and Media ResourcesWhen your network is experiencing codec or CAC problems, calls are failing, and media resources are being used perhaps unnecessarily. This leads to resources not being available when needed.

You can resolve these issues as follows:

■ CAC and codec: Check the device pool and make sure that it matches the region and locations of the new site.

■ Media resources: Verify MRGL mappings in the device pool.

SRSTRoaming phones are not registering to sites’ SRST when isolated from UCM. To resolve this issue, verify SRST under the site-specific device pool.

DHCP and iPThese problems are old problems that affect all phones, and even Device Mobility. To resolve these issues, follow IP steps as mentioned earlier in this guide. Check the device pool. The device pool should have the correct Device Mobility group and location configured.


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Section 14 Troubleshooting Device Mobility Issues

Call Routing and AARWhen calls are being sent to the wrong gateways, AAR does not work, and localization displays the incorrect format. These are just a couple of the issues that can happen. The failure of call routing and AAR are going to be design or configuration related.

Device Mobility Call Routing Problems■ CSS misconfiguration

■ Incorrect Device Mobility

■ Inaccurate CSS design

■ Incorrect configuration of device pools

Ways to resolve these issues include the following:

■ View the current phone Device Mobility settings.

■ Trace and use DNA to check the dial plan.

■ Confirm partition and CSS configurations.

■ Confirm the device pool configuration.

■ Check the Device Mobility group design.


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Section 15

Extension Mobility Issues

Extension Mobility enables users to log in to almost any phone and bring their personal and customized phone settings with them. The same basic components are used; they are just made in different fashion. Consider it like making a peanut butter and jelly sandwich. Depending on the person making the sandwich, or where you’re located geographically, will determine if something is added. So, when it comes to Extension Mobility, you still have the same building blocks (phone button templates, speed dial, service URLs, MOH, and so on).

Now your sandwich has cream cheese and marshmallows.

Mobility problems include the following:

■ CoS

■ Call routing

■ Assorted login and logout issues

■ Phone service

■ Phone buttons

Mobility Error MessagesCisco IP phones should display some type of message on the phone screen. Here are some Extension Mobility error messages you might see:


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Section 15 Extension Mobility Issues

■ Host not found: Check that the Cisco Tomcat service is running (Serviceability, Control Center, Network Services).

■ [6]-Database Error: Check to see whether large number of database request exists. (RTMT, Requests in Progress counter in the Cisco Extension Mobility object).

■ [201]-Authentication error: Make sure the user ID and PIN were entered correctly in Unified Communication Manager (User Management, end user).

■ [202]-Blank userid or pin: Enter a valid user ID or PIN.

■ [205]-User Profile Absent: Check that user has a device profile associated (User Management, end user).

■ [207]-Device name Empty: Check that Cisco Extension Mobility has the correct URL configured.

■ [208]-EM Service Conn error: Check that the Extension Mobility service is running (Serviceability, Control Center, Feature Services).

■ [22]-Dev. Logon disabled: Verify that Enable Extension Mobility is checked on the phone’s configuration page.

■ [25]-User logged in elsewhere: Self-explanatory.

■ [26]-Busy, please try again: Verify that the number of concurrent login and logout requests is not larger than allowed (RTMT, Requests in Progress counter in the Cisco Extension Mobility object).

■ Http Error [503]: Check that the Cisco IP Phone Services service is running (Serviceability, Control Center, Feature Services).

Mobility login issues include the following:

■ Service not available: Phone is using either a logout-device profile or phone-device profile that doesn’t subscribe to the Cisco Extension Mobility Service. Solution: Check that the device profile used as the logout profile on the phone subscribes to Extension Mobility.

■ User cannot login: Users are having problems logging in to phones. Solution 1: Verify username and passwords are correct.


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Solution 2: associated user with a device profile for Extension Mobility.

If the username, password, and profile are correct, try Solution 3.

Solution 3: Check the phone configuration page and verify that Enable Extension Mobility is checked.

■ Phone restarts: User can log in fine, but the phone restarts instead of resetting. Solution: Make sure locales match on the phone, end-user configuration page, and the device profile.

Logging OutThe same issues that affect the user logging in can cause the problems logging out. To resolve logging-out issues, follow logging-in troubleshooting suggestions.

Extension Mobility Phone Button issuesThis will more than likely be caused by a mismatch with phone models. You can create different profiles per phone type.

Extension Mobility Calling and Call Routing issuesThe key points are as follows:

■ Device CSS is never modified by Extension Mobility.

■ Line CSS is changed by Extension Mobility.

■ When using local route groups, the line or device CSS approach

Applies the line CSS of the user device profile, and user CoS settings are enforced.


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Device CSS is not modified.

AAR CSS is never changed by Extension Mobility.

In a traditional CSS approach, the same CSS is used, causing problems with users’ CoS in multisite deployments.

With these key points in mind, you can start to resolve Extension Mobility calling and call routing issues, just as with other calling issues previously mentioned.


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Section 16

Cisco Unified Mobility General Issues

New concepts are associated with mobile voice access, and so you might face the following issues:

■ Users unable to switch calls between desk phones and mobile phones.

■ Mobile voice access unavailable or not allowing outbound calls.

■ Call transfers, resume, and call park or handing off a call not working on a remote phone.

■ Internal called party unreachable.

■ Incorrect number presented.

■ Mobile or desktop phones not ringing on inbound calls.

Unified Mobility Mobile Connect Troubleshooting■ The Mobility button is not available. Solution: Check that the correct softkey template is used for the phone.

■ An error message displays when the Mobility button is used. Solution: Make sure the phone is associated with a Cisco Unified Mobility end user.

■ Inbound calls to an office phone rings the office phone but not an associated remote phone. Solutions include the following:

Enable Mobile Connect.


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Section 16 Cisco Unified Mobility General Issues

Check whether the ring schedule is configured. Calls falling outside of the ring schedule will not ring the phone.

Enable Mobile Connect for Remote Destination.

Associate the remote destination profile with the end user.

Associate the remote destination line with the remote destination profile.

Check CSS configurations.

Check the access list.

Remote Phone Calling issuesWhen the wrong number (caller ID) is presented while using a remote phone, several issues are the likely causes. Solutions include the following:

■ Check that the remote destination line is associated with the office phone.

■ Check that the remote destination profile line is in the same partition as the office phone.

■ Check that Mobile Connect is enabled for the end user.

■ If all the preceding troubleshooting attempts fail, check the UCM service parameters to see how caller ID matching is set.

Redirection to Remote Phone issuesA user cannot redirect calls to a remote phone from an office phone. Solutions include the following:

■ Same four troubleshooting tips listed with regard to remote phone calling issues.

■ The Mobility softkey button is missing or the template is missing the Mobility button.

■ CSS issues.

■ UCM dial plan, localization, or globalization is not working correctly.


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Section 16 Cisco Unified Mobility General Issues

Mobility Mobile Voice Access issues■ Users not able to reach interactive voice response (IVR). Solution: Misconfigured IVR or problem with inbound dial peer.

■ Users unable to reach destination number after dialing. Solution: Check the calling number digit manipulation on the gateway.

■ Incorrect configuration of inbound dialing on the gateway. Solution: Check CSS of the user.

Mobility Mobile Voice Access Outgoing Call issues■ User can call into the IVR, get authenticated, and dial the destination number, but the call fails. Solution: Check and recheck

CSS.


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Section 17

Troubleshooting Media Resources

Music on Hold (MOH)When a call is placed on hold, you can choose to play music or informational messages. The MOH can be either multicast or unicast, depending on the requirements of your company.

Common MOH issues include the following:

■ MOH resources not available. Solution: Use RTMT performance counters to view MOH resources.

■ MOH registration problems. Solution: If your MOH resources are not available, one area of concern might be some MOH resources are not registering:

Network connectivity

QoS

Database corruption

MOH resource disabled

Cisco IP Voice Media Streaming Service not running

■ The call is disconnected when placed on hold. Solution: Open Cisco IP Voice Media Streaming Application Service and restart. Set the codec in System, Service Parameters. Then, restart the Cisco Voice Media Stream app service in Control Center, Feature Services.


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Section 17 Troubleshooting Media Resources

■ Audio too silent or loud. Solution: Change the volume with service parameters in the Cisco IP Voice Media Streaming Application Service, and then restart the service. Changes will affect only newly imported files. Re-import wanted files to the MOH server.

■ TOH (Tone on Hold). Solution: Possible mismatch in codec configuration between the region and the MOH server.

Location based CAC is used, and not enough bandwidth is available.

UCM will tell phone to generate TOH.

No more audio streams are available for the MOH server, so TOH is used.

The MOH server is not registered.

The MRG configuration is incorrect.

■ Multicast MOH is expected, but unicast MOH is played. Solution: Multicast MOH is not enabled.

Multicast max hops is set too low.

The Cisco IOS router is not enabled for multicast routing.

IOS commands to deal with this issue include the following:

ip multicast-routing enables multicast routing.

ip pim enables the interface to participate in multicast.

ip pim sparse-dense-mode enables the interface to participate.

Note that you use only one ip pim command, depending on your needs.

You can verify multicast routing in IOS with the following commands:

show ip multicast shows multicast status.

show ip mroute displays the multicast routing table.

show ip multicast interface interface shows interface configuration and packet counters


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Section 17 Troubleshooting Media Resources

To verify multicast in UCM, use traces.

■ A remote MOH source is used, when a local MOH source is available.


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Section 18

Troubleshooting Media Termination Points

A Media Termination Point (MTP) software device allows supplementary services, such as call park, call hold, call transfer, and conferencing, that are otherwise not available when calls are routed to an H.323 endpoint. Some H.323 gateways might require that calls use an MTP to enable supplementary call services, but Cisco IOS gateways do not, as a general rule.

MTP Registration issuesCommon MTP registration issues include the following:

■ Cisco IP Voice Media Streaming Application is not running properly. Solution: Restart the service in Unified Serviceability.

■ Hardware MTP names not matching with Unified Communications Manager. Solution: Check the configuration.

Perform no shutdown in the farm profile service parameters.

■ Access list filtering. Solution: Allow TCP port 2000.

■ Software MTP not activated. Solution: Activate MTP in the Cisco IP Voice Media Streaming Service application.

MTP Allocation issues■ MTP not registered with Cisco Unified Communications Manager. Solution: Register MTP with UCM. See the registration

troubleshooting section.


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Section 18 Troubleshooting Media Termination Points

■ MTP is running out of resources.Solution: Deploy more RTP resources if they are run out frequently.

■ MRG and MRGL misconfigured. Solution: Check your MRG and MRGL configurations. Solution: Associate MGRL with a device that requires MTP.

Verifying MTPYou should run a trace and check output:

debug sccp messages shows SCCP messages.show sccp connections shows sessions and both legs.


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Section 19

Troubleshooting Conferences

Conference calls enable three or more people to participate in the same phone conversation. Conference calls are a popular way for companies to save money on travel-related expenses. In the world of Cisco, you have two types of conference bridges: software and hardware. Each has its advantages and disadvantages. The conference can be either Meet-Me or ad hoc.

Verify Hardware■ Cisco IOS: show sccp displays the status and the IP address of the UCM it is registered with.

■ Cisco IOS: show sccp ccm group number: displays the name used to register with UCM.

■ UCM: Media Resources, Conference Bridge.

Verify SoftwareYou want to confirm that the Cisco IP Voice Media Streaming Application Service is running: Serviceability, Tools, Control

Center, Feature Services.

Ad Hoc Conferencing issuesConference participants cannot be added by the originator:

■ No Confrn softkey is on the phone. Solution: Check the phone configuration page and make sure the softkey is added.

■ No Conference Bridge message appears on the originator’s phone. Solution: Associate the phone with MRGL.


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Section 19 Troubleshooting Conferences

■ Maximum number of participants reached. Solution: Check Maximum Ad Hoc Conference service parameters.

■ Cisco IP Voice Media Streaming Application Service is not running. Solution: Enable the service in Serviceability, Tool, Control Center, Feature Services.

■ Conference bridges out of resources. Solution: Add more resources.

■ Conference participants cannot be added by the originator. Solution 1: Same as Steps 1–6. Solution 2: Check whether Advanced Ad Hoc is enabled.

■ Conference established, but suddenly disconnects. Solution: Check the Drop Ad Hoc settings:

Never: Conference remains active if the originator leaves.

When Conference Controller Leaves: Terminates the conference if the originator leaves, transfers, parks, or redirects the call.

When No On-Net Parties Remain in Conference: Terminates when no on-net parties remain.

PerformanceUse RTMT to monitor the performance of your conference resources.

TracingTraces can provide lots of information about the conference.


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Section 19 Troubleshooting Conferences

Meet-Me Conferencing issues■ The Meet-Me softkey missing. Solution: Check the softkey template.

■ The Meet-Me pattern to join a conference presents a reorder tone. Solution: Check to see whether the conference has been set up.

Check the caller CSS.

Check the codec of the caller.

■ Conference bridge resources are completely used up. Solution: Check the value of the Maximum Meet-Me Conference unicast service parameter in UCM.

■ No conference bridge is displayed and a reorder tone occurs when setting up a new conference. Solution: Associate the phone with a conference bridge. Check MRG and MGL for misconfigurations.


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Section 20

Troubleshooting Transcoder Issues

Transcoders allow communication between endpoints that use different codecs. The transcoder converts the incoming streams to the other codec and outputs them to the endpoints.

Transcoder RegistrationFor transcoders to work, they must be correctly registered. You should check Cisco IOS and your Cisco Unified Communication Manager settings:

■ IOS: Make sure that the name matches by issuing the command show sccp ccm group number. Then, verify that the transcoder is not shut down. You should issue the no shutdown command.

■ UCM: The Cisco IP Voice Media Streaming Application Service might not be operating properly. Restart the service in Unified Serviceability.

■ Firewall: The firewalls must allow TCP port 2000 (SCCP) to pass through.

Transcoder Allocation issues■ Check that the transcoding resource is registered with UCM.

■ Transcoder out of resources. Solution: Look at deploying more transcoder resources.

■ MRG/MRGL misconfigured. Solution: Check the configurations in UCM.


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Section 20 Troubleshooting Transcoder Issues

Verifying Transcoder ResourcesYou can verify transcoding resources with RTMT:

show sccp connections shows sessions and both legs.show dspfarm dsp all shows which digital signal processor (DSP) resources are performing transcoding and which are idle.

Tracing is possible and provides useful information.

Debuggingdebug sccp messages shows SCCP messages sent and received, IP addresses and ports, and so on.


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© 2011 Cisco Systems Inc. All rights reserved. This publication is protected by copyright.


CCNP Voice TVoice 642-427 Quick ReferenceBrion S. Washington

Technical Reviewer: Alex Hannah

Copyright © 2011 Pearson Education, Inc.Published by:Cisco Press800 East 96th Street Indianapolis, IN 46240 USAAll rights reserved. No part of this digital Quick Reference may be reproduced or transmitted in any form or by any means, electronic or mechanical, including photocopying, recording, or by any information storage and retrieval system, without written permission from the publisher, except for the inclusion of brief quotations in a review.Digital Edition April 2011ISBN-10: 0-13-249816-2ISBN-13: 978-0-13-249816-6

Warning and DisclaimerThis digital Quick Reference is designed to provide information about the CCNP Voice TVoice 642-427 exam. Every effort has been made to make this digital Quick Reference as complete and accurate as possible, but no warranty or fitness is implied.The information is provided on an “as is” basis. The authors, Cisco Press, and Cisco Systems, Inc., shall have neither liability nor responsibility to any person or entity with respect to any loss or damages arising from the information contained in this digital Quick Reference.The opinions expressed in this digital Quick Reference belong to the authors and are not necessarily those of Cisco Systems, Inc.

Trademark AcknowledgmentsAll terms mentioned in this digital Quick Reference that are known to be trademarks or service marks have been appropri-ately capitalized. Cisco Press or Cisco Systems, Inc., cannot attest to the accuracy of this information. Use of a term in this digital Quick Reference should not be regarded as affecting the validity of any trademark or service mark.

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Cisco has more than 200 offices worldwide. Addresses, phone numbers, and fax numbers are listed on the Cisco Website at www.cisco.com/go/offices.

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All other trademarks mentioned in this document or website are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (0812R)

Cisco has more than 200 offices worldwide. Addresses, phone numbers, and fax numbers are listed on the Cisco Website at www.cisco.com/go/offices.

CCDE, CCENT, Cisco Eos, Cisco HealthPresence, the Cisco logo, Cisco Lumin, Cisco Nexus, Cisco StadiumVision, Cisco TelePresence, Cisco WebEx, DCE, and Welcome to the Human Network are trademarks; Changing the Way We Work, Live, Play, and Learn and Cisco Store are service marks; and Access Registrar, Aironet, AsyncOS, Bringing the Meeting To You, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, CCVP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Collaboration Without Limitation, EtherFast, EtherSwitch, Event Center, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iPhone, iQuick Study, IronPort, the IronPort logo, LightStream, Linksys, MediaTone, MeetingPlace, MeetingPlace Chime Sound, MGX, Networkers, Networking Academy, Network Registrar, PCNow, PIX, PowerPanels, ProConnect, ScriptShare, SenderBase, SMARTnet, Spectrum Expert, StackWise, The Fastest Way to Increase Your Internet Quotient, TransPath, WebEx, and the WebEx logo are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.

All other trademarks mentioned in this document or website are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (0812R)

Americas HeadquartersCisco Systems, Inc.San Jose, CA

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Americas HeadquartersCisco Systems, Inc.San Jose, CA

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www.cisco.com/go/offices

Documents

CCNP Voice TVoice 642-427 Quick Referenceebook.konfigurasi.net/Cisco/CCNP-Voice-Tvoice.pdf · This Quick Reference is a final preparation resource. With the assistance of this guide