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Corporate Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA http://www.cisco.com Tel: 408 526-4000 800 553-NETS (6387) Fax: 408 526-4100 Cisco IOS Voice Troubleshooting and Monitoring Guide Release 12.4

Troubleshooting and Monitoring Voice

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Troubleshooting Cisco Voice

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  • Corporate HeadCisco Systems, In170 West TasmanSan Jose, CA 951USAhttp://www.cisco.Tel: 408 526-400

    800 553-NEFax: 408 526-410

    Cisco IOSMonitorinRelease 12.4 Voice Troubleshooting and g Guidequartersc. Drive34-1706

    com0TS (6387)0

  • THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.

    THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.

    The Cisco implementation of TCP header compression is an adaptation of a program developed by the University of California, Berkeley (UCB) as part of UCBs public domain version of the UNIX operating system. All rights reserved. Copyright 1981, Regents of the University of California.

    NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED AS IS WITH ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE.

    IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.

    Cisco IOS Voice TroubleshoCopyright 2004, 2005 Ci

    CCSP, CCVP, the Cisco SquiQuick Study are service marCertified Internetwork ExperEnterprise/Solver, EtherChanNet Readiness Scorecard, LiProConnect, RateMUX, Scritrademarks of Cisco Systems

    All other trademarks mentionbetween Cisco and any otheroting and Monitoring Guidesco Systems, Inc. All rights reserved.

    are Bridge logo, Follow Me Browsing, and StackWise are trademarks of Cisco Systems, Inc.; Changing the Way We Work, Live, Play, and Learn, and ks of Cisco Systems, Inc.; and Access Registrar, Aironet, ASIST, BPX, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, Cisco, the Cisco t logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Empowering the Internet Generation, nel, EtherFast, EtherSwitch, Fast Step, FormShare, GigaDrive, GigaStack, HomeLink, Internet Quotient, IOS, IP/TV, iQ Expertise, the iQ logo, iQ

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  • C O N T E N T S

    About Cisco IOS Software Documentation xxi

    Documentation Objectives xxi

    Audience xxi

    Documentation Organization xxiSupporting Documents and Resources xxii

    Document Conventions xxiv

    P A R T 1iiiCisco IOS Voice Troubleshooting and Monitoring Guide

    Obtaining Documentation xxvCisco.com xxvOrdering Documentation xxvi

    Documentation Feedback xxvi

    Obtaining Technical Assistance xxviCisco Technical Support Website xxviiSubmitting a Service Request xxviiDefinitions of Service Request Severity xxvii

    Obtaining Additional Publications and Information xxviii

    Guide to Cisco IOS Voice Troubleshooting and Monitoring Features xxix

    Cisco IOS Release 12.4(4)T xxx

    Cisco IOS Release 12.3(8)T xxx

    Cisco IOS Release 12.3(4)T xxxi

    Cisco IOS Release 12.2(11)T xxxi

    Troubleshooting Cisco IOS Voice Overview

    Voice Call Flow Overview 3

    Call Setup 3Call Setup Elements 4

    Source and Destination POTS Dial Peers 4Voice Network Dial Peers 4Inbound and Outbound Call Legs 5

    Call Setup Process 6Dial Peer Matching 7

    Inbound Dial Peer Matching 8Outbound Dial Peer Matching 11

  • Contents

    ivCisco IO

    Voice Network Dial Peer Matching 13POTS Dial Peer Matching 13

    Call Flow Through Router Components 14Telephony Interface Architecture 15Voice Application Interface 16

    Debug Command Output on Cisco IOS Voice Gateways 17

    Voice Debug Concepts 17Debug Header Format 18

    IVR Module-Dependent List 19CallEntry ID and GUID Call Legs 20Enabling Command Profile Debugging 20S Voice Troubleshooting and Monitoring Guide

    Fax Debug Profile 21Modem Debug Profile 23Voice Debug Profile 24

    Enabling Individual Debug Commands for CCAPI, TSP, and VTSP Debugging 26

    Managing the Voice Call Debug Output 35Supported Commands 35

    Sample Output Examples for the Enhanced debug Commands 38Full GUID Header: Example 38Short Header: Example 39Setup and Teardown: Example 39

    Filtering Troubleshooting Output 49

    Filtering Output from show and more Commands 49

    Voice Call Debug Filtering on Cisco Voice Gateways 50Restrictions for Voice Call Debug Filtering 50Information About Voice Call Debug Filtering 50

    Debug Commands that Support Voice Call Filtering 51Generic Call Filter Module 52Calling and Called Number Strings 53Exact and Partial Matching 55Media and Signaling Streams 55

    Configuring the Voice Call Debug Filter 55Configuring Call-Specific Conditions 55Enabling Debug for the Set Filtering Conditions 59

    Output Examples for Voice Call Debug Filtering 62Exact Match Filtering: Example 62Partial Match Filtering: Example 65

    Voice Call Debug Filtering on H.323 Gatekeepers 69

  • Contents

    Restrictions for Voice Call Debug Filtering on Voice Gatekeepers 69Information About Voice Call Debug Filtering 69

    Debug Commands that Support Voice Call Filtering on Cisco Voice Gatekeepers 70Gatekeeper Filter Module 70Calling and Called Number Strings 71Exact and Partial Matching Conditions Rules and Information 71

    Configuring the Voice Call Debug Filter for Cisco Gatekeepers 79Configuring Call-Specific Conditions for Gatekeepers 79Enabling Debug for the Set Filtering Conditions 80Prerequisites 80Troubleshooting Tips 81vCisco IOS Voice Troubleshooting and Monitoring Guide

    Examples 82

    SIP Debug Output Filtering Support 83Restrictions for SIP Debug Output Filtering Support 83Information About SIP Debug Output Filtering Support 83

    Feature Design of SIP Debug Output Filtering Support 83Generic Call Filtering Module 84SIP Debug Commands that Support Output Filtering 84Matching Conditions 84

    Configuring SIP Debug Filtering 85Configuring Call Filters 85Enabling SIP Debug Output Filtering 87Verifying SIP Debug Output Filtering Support 89

    Configuration Examples for SIP Debug Filtering 90Configuring Call Filters: Example 90Enabling SIP Debug Output Filtering: Example 94

    MGCP Call Centric Debug 96Restrictions for MGCP Call Centric Debug 96Information About MGCP Call Centric Debug 96

    MGCP Debug Commands that Support Debug Filtering 97Match Conditions for MGCP Debug Filtering 97Trace Levels for MGCP Debug Output 98Tips on Collecting Debug Output 98

    How to Enable MGCP Call Centric Debug 99Modifying the Debug Header Format for MGCP Debug Output 99Creating Match Lists for MGCP Filtering Conditions 100Enabling MGCP Debug Filtering Using Match Lists 102Prerequisites 102Restrictions 102Verifying the MGCP Debug Filtering Configuration 103

  • Contents

    viCisco IO

    Enabling MGCP Debug Trace Levels 104Restrictions 104

    Configuration Examples for MGCP Call Centric Debug 105Match-List Configuration for MGCP Debug Filtering: Example 105Enabling MGCP Debug Filtering: Example 108

    Cisco VoIP Internal Error Codes 109

    Prerequisites for Cisco VoIP Internal Error Codes 109

    Restrictions for Cisco VoIP Internal Error Codes 109

    Information About Cisco VoIP Internal Error Codes 109Benefits of Cisco VoIP Internal Error Codes 110S Voice Troubleshooting and Monitoring Guide

    Feature Design of Cisco VoIP Internal Error Codes 110IEC Reporting 110

    Gatekeeper Behavior and Cisco VoIP Internal Error Codes 111Obtaining IECs 113

    Internal Error Code Notation 115Entity 115Category Codes 116Subsystem Codes 117Error Codes 118

    How to Configure IEC Options 146Configuring IEC Options 146

    Enabling IEC Syslog Reporting 146Configuring Cause Code Mapping 147Troubleshooting Tips 148What to Do Next 148

    New and Modified Configuration Commands for Gatekeeper IECs in Cisco IOS Release 12.4(4)T 148Verifying IEC Options 149

    Prerequisites 149Displaying IEC Options 150

    Configuration Examples for Cisco VoIP Internal Error Codes 151Enabling IEC Syslog Reporting and Configuring Cause Code Mapping: Example 152Verifying IEC Configuration: Example 152

    Sample Output from the show running-config Command: Example 152Sample Output from the show voice iec description Command: Example 154Sample Output from the show voice statistics iec Command: Example 154Sample Output from the clear voice statistics Command: Example 156Sample Output from the show voice cause-code category-q850 Command: Example 156

    Troubleshooting VoIP Networks Using Cisco VoIP Internal Error Codes 157

  • Contents

    Troubleshooting Two-Stage Dialing Failures 157Symptom 157Problem Description 157Troubleshooting Tasks 157

    Troubleshooting Socket Failures 161Symptom 161Problem Description 161Troubleshooting Tasks 161

    Troubleshooting Resources 171

    Troubleshooting Tools 171Using Router Diagnostic Commands 171

    P A R T 2viiCisco IOS Voice Troubleshooting and Monitoring Guide

    Using show Commands 171Using debug Commands 172Using the ping Commands 173Using the trace Commands 174

    Using Cisco Network Management Tools 174CiscoView 175Internetwork Performance Monitor 175TrafficDirector RMON Application 175VlanDirector Switch Management Application 176

    Third-Party Troubleshooting Tools 176Volt-Ohm Meters, Digital Multimeters, and Cable Testers 177TDRs and OTDRs 177Breakout Boxes, Fox Boxes, BERTs and BLERTs 177Network Monitors 178Network Analyzers 178

    Other Troubleshooting Resources 178Internetworking Troubleshooting Handbook 178TAC Troubleshooting Tools on Cisco.com 179

    Troubleshooting Cisco IOS Voice Telephony

    Troubleshooting Analog Voice Interfaces to the IP Network 183

    FXS Interfaces 183FXS Hardware Troubleshooting 184

    Software Compatibility 184Cabling 184Shutdown Port 185Disabling a Port on a Multiple Port Card 186

  • Contents

    viiiCisco IO

    Ring Voltage Problems 186Ringing Voltages 186Idle Battery Voltage 186Idle Line Voltages 187Ring Voltage Problems 187

    Unbreakable Dial Tone 188No LED When Phone Off the Hook 189

    FXO Interfaces 189FXO Hardware Troubleshooting 190

    Software Compatibility 190Cabling 190S Voice Troubleshooting and Monitoring Guide

    Shutdown Port 191Disabling a Port on a Multiple Port Card 192

    FXO Disconnect Failure 192Troubleshooting FXO Answer and Disconnect Supervision 192

    Monitoring and Maintaining FXO Answer and Disconnect Supervision 194Unbreakable Dial Tone 194Troubleshooting Caller ID Problems 194

    Calling Number Lost, Name Delivered 195Calling Number Delivered, Name Lost 195

    E&M Interfaces 196E&M Hardware Troubleshooting 197

    Software Compatibility 197Cabling 197Shutdown Port 198

    E&M Interface Types 199E&M Signaling Unit Side and Trunk Circuit Side Compatibility Issues 199E&M Type I Interface Model 200E&M Type II Interface Model 202E&M Type III Interface Model 204E&M Type V Interface Model 206

    Troubleshooting E&M Interfaces at the Physical Level 208Preparing to Troubleshoot E&M Physical Problems 209Troubleshooting Type I Interfaces 212Troubleshooting Type II Interfaces 213Troubleshooting Type III Interfaces 214Troubleshooting Type V Interfaces 215

    Confirming E&M Configuration 216Confirming the PBX E&M Configuration Parameters 216Confirming the Cisco IOS Gateway Configuration 216

  • Contents

    Verifying the Wiring Arrangement Between the PBX and the Cisco Gateway 217Verifying Supervision Signaling 218Verifying That the Cisco Equipment and PBX Are Sending and Receiving Digits 219Verifyinng That the Gateway Sends the Expected Digits to the PBX 222Verify That the Gateway Receives the Expected Digits from the PBX 223

    Unbreakable Dial Tone 223

    Analog DID Interfaces 224DID Hardware Troubleshooting 225

    Software Compatibility 225Cabling 225Shutdown Port 225ixCisco IOS Voice Troubleshooting and Monitoring Guide

    Verifying Direct Inward Dialing Voice-Port Configuration 226

    Voice Port Testing Commands 227Detector-Related Function Tests 227Loopback Function Tests 228Tone Injection Tests 228 Relay-Related Function Tests 229 Fax/Voice Mode Tests 229

    Troubleshooting Digital Voice Interfaces to the IP Network 231

    Checking the Hardware 231Software Compatibility 232Cabling 232

    T1/E1 Trunk and Digital Voice Port Pinouts (RJ-48) 232T1/E1 Trunk and Digital Voice Port Pinouts (RJ-45) 233

    Shutdown Port 233

    Checking the Digital Signal Processors 233Voice DSP Control Message Logger 234

    Message Logger Overview 234Configuration Tasks 235Configuration Examples 238

    Voice Call Tuning 239Voice DSP Crash Dump File Analysis 239

    How to Configure Voice DSP Crash Dump File Analysis 240Troubleshooting Voice DSP Crash Dump File Analysis 241Verifying DSP Crash Dump File Analysis 242Configuration Examples for Voice DSP Crash Dump File Analysis 242

    Troubleshooting Universal Port SPEs 243Configure SPE Diagnostic Tests 243SPE Disconnect Reason Codes 245

  • Contents

    xCisco IO

    DSP Troubleshooting Links 260

    Verifying Codec Complexity 260Codec Complexity Mismatch 260

    Checking the Interface 261Troubleshooting T1 and E1 Layer 1 Problems 262

    Controller Is Administratively Down 262Controller Has Loss of Frame 263Controller Has Loss of Signal 264

    Checking T1/E1 Controller Configuration 265Framing Formats on Digital T1/E1 Voice Ports 266Clock Sources on Digital T1/E1 Voice Ports 266S Voice Troubleshooting and Monitoring Guide

    Line Coding on Digital T1/E1 Voice Ports 268T1/E1 Channel-Associated Signaling 269

    Troubleshooting Commands 270E1 R2 Interfaces 270

    Troubleshooting E1 R2 Failures 270debug and show Commands 271

    ISDN Interfaces 272ISDN PRI Troubleshooting Tips 273Verifying the ISDN Switch Type and PRI Group Timeslot Configuration 274Ringback 275QSIG Protocol Support 276

    Troubleshooting Drop-and-Insert 277Troubleshooting Transparent Common Channel Signaling 278Dial Tone Issues 278

    Router Does Not Recognize Voice Port 278Voice Ports Are in the Shutdown State 279Voice Ports Configured as Connection Trunk 279No Dial Tone on Digital Voice Port 279Debug Command Output Shows VTSP Timeout 279

    Verifying Digital Voice-Port Configurations 280show voice port Samples 281show controller Samples 283show voice dsp Samples 283show voice call summary Samples 284show call active voice Samples 285show call history voice Sample 286Verifying Digits Received and Sent on the POTS Call Leg 287

    Voice Port Testing Commands 289

  • Contents

    Detector-Related Function Tests 290Loopback Function Tests 290Tone Injection Tests 290Relay-Related Function Tests 291Fax/Voice Mode Tests 291

    Troubleshooting Quality of Service for VoIP 293

    Understanding VoIP QoS Issues 293Measuring QoS 294

    Common QoS Problems 295Delay in Voice Networks 296Packet Loss 298

    P A R T 3xiCisco IOS Voice Troubleshooting and Monitoring Guide

    Jitter Adjustment 298Echo Adjustment 301

    Determine Where Echo Is Occurring 301Troubleshooting Echo in Cisco IOS Gateways 302Measuring Echo in Cisco IOS Gateways 303

    Voice Level Adjustment 306Input and Output Levels 306Voice Activity Detection Level 308

    Voice Call Tuning 309Restrictions for Voice Call Tuning 309Information About Voice Call Tuning 309

    Packet-Flow Detection 309DSP State 310Echo-Cancellation State 310Jitter-Buffer Parameters 310

    How to Verify Voice Call Tuning Functionality 310Voice Call Tuning Configuration Examples 311

    Troubleshooting Cisco IOS Voice Technologies

    Troubleshooting H.323 Interfaces to the IP Network 315

    H.323-Related Standards 316H.225 Signaling 317H.245 Signaling 318

    Troubleshooting Gateways 319Troubleshooting H.323 Gateway Call Routing and Dial Peers 319

    Verifying Digits Received and Sent on the POTS Call Leg 319Verifying End-to-End VoIP Signaling on the VoIP Call Leg 322

  • Contents

    xiiCisco IO

    Troubleshooting H.323 Gateway Dial Tone 332Troubleshooting H.323 Gateway Busy Tone 332

    No DTMF Digits or Audio Passed on VoIP Calls to PSTN or PBX 333No Busy Tone or Announcement Message Received When Placing VoIP Outbound Calls 333No Busy Tone on Inbound Call from Telephony (ISDN) to Cisco CallManager IP Phone, Cisco IOS Gateway, or Third-Party H.323 Device 333

    Troubleshooting H.323 Gateway Ringback 334No Ringback Tone on VoIP Toll-Bypass Calls 334No Ringback Tone on VoIP Inbound Calls to Cisco CallManager (or Third-Party VoIP Devices) Through Cisco IOS Gateway 335No Ringback Tone on VoIP Outbound Calls from Cisco CallManager (or Third-Party Device) Through Cisco IOS Gateway 336S Voice Troubleshooting and Monitoring Guide

    No Ringback to PSTN When IP Phones Initiate a Call Transfer (Cisco CallManager or Cisco Unity Voice Mail) 337

    Troubleshooting H.323 Gateway One-Way or No Audio 338Ensuring IP Routing Is Enabled on Cisco IOS Gateways 338Checking Basic IP Routing 339Binding the H.323 Signaling to a Specific IP Address 339Checking That Answer Supervision Is Being Sent and Received Correctly From the Telco or Switch 340Using the voice rtp send-recv Command to Establish Early Two-Way Audio 340Checking cRTP Settings on a Link-by-Link Basis 340Verifying Minimum Software Level for NAT on Cisco IOS Gateways 341Disabling voice-fastpath on Cisco AS5350 and Cisco AS5400 Universal Gateways 341Configurinng the VPN IP Address with SoftPhone 341Verifying One-Way Audio 342

    Using the Test Call Feature to Verify Voice Path 342Information About the Test Call Feature 343Limitations 343Test Call Command-Line Interface 344Sample Tasks 346

    Troubleshooting Gatekeepers 347Troubleshooting H.323 Gatekeeper Registration 348

    Related Commands 348Reject Reasons 350

    Troubleshooting H.323 Gatekeeper Call Routing and Dial Peers 352Troubleshooting H.323 Gatekeeper Bandwidth 353

    Bandwidth Management Operation Overview 353Configuring the Bandwidth Management Feature on the Cisco Gatekeeper 354Using Gatekeeper show Commands to Display Bandwidth Information 354Bandwidth-Related RAS Messages (BRQ, BCF, and BRJ) 355

  • Contents

    Checking Cisco Gateway Failover to Alternate Gatekeeper 356Gatekeeper Update Protocol 357

    Troubleshooting Issues with Alternate Endpoints 358Troubleshooting Gatekeeper Endpoint Call Admission Issues 359Troubleshoot Load Balancing 365

    Gateway-to-Gateway and Gatekeeper-to-Gateway Security 365

    Troubleshooting SIP Interfaces to the IP Network 367

    Troubleshooting the Cisco SIP IP Phone 7960 367Troubleshooting Features 367Troubleshooting Tips 368

    Cisco SIP IP Phone Is Unprovisioned or Is Unable to Obtain an IP Address 369xiiiCisco IOS Voice Troubleshooting and Monitoring Guide

    Cisco SIP IP Phone Does Not Register With the SIP Proxy or SIP Registrar Server 369Outbound Calls Cannot Be Placed from a Cisco SIP IP Phone 370Inbound Calls Cannot Be Received on a Cisco SIP IP Phone 370Poor Voice Quality on the Cisco SIP IP Phone 370DTMF Digits Do Not Function Properly 371Cisco SIP IP Phones Do Not Work When Plugged into a Line-Powered Switch 371Call Transfer Does Not Work Correctly 371Some SIP Messages are Retransmitted Too Often 371

    Troubleshooting the Cisco SIP Gateway 371Unable to Make Outbound Calls from the Cisco SIP Gateway to a SIP Endpoint 372Unable to Make Inbound Calls to a PSTN Through a Cisco SIP Gateway 372Calls to a PSTN via the Cisco SIP Gateway Fail with a 400 Bad Request Response 373Voice Quality Is Compromised on Calls Through or From the Cisco SIP Gateway 376Some SIP Messages Are Retransmitted Too Often 376Call Transfer Does Not Work Correctly 376Troubleshooting Commands 377

    Troubleshooting the Cisco SIP Proxy Server 386Troubleshooting Tips 387Cisco SIP Proxy Server Does Not Start 387Cisco SIP Proxy Server Does Not Allow Devices to Register 387Cisco SIP Proxy Server Does Not Route Calls Properly 388Cisco SIP Proxy Server Reports That SIP Messages Are Bad 388Cisco SIP Proxy Server Farming Does Not Work Correctly 388Voice Quality Problems 388

    SIP Messages and Methods 388Requests 389Responses 389Registration Process 389

  • Contents

    xivCisco IO

    Invitation Process 389

    Troubleshooting MGCP and Related Protocol Interfaces to the IP Network 391

    MGCP Overview 391MGCP Gateway Call Flow 393

    User Access Verification 393Troubleshooting Guidelines 397

    Call Routing and Dial Peers 398Verifying Digits Received and Sent on the POTS Call Leg 399

    show dialplan number 399debug vtsp dsp 400

    Verifying End-to-End VoIP Signaling on the VoIP Call Leg 401S Voice Troubleshooting and Monitoring Guide

    Call Admission Control 401Troubleshooting MGCP 402Troubleshooting MGCP SRC CAC 402Troubleshooting MGCP RSVP CAC 403Troubleshooting MGCP SA Agent CAC 403

    Verifying Connections and Endpoints 404

    MGCP Testing Commands 405show ccm-manager 405show mgcp 406show mgcp endpoint 407show mgcp connection 407show voice port mod_num/slot_num/port_num 409show mgcp statistics 412Other debug mgcp Commands 413

    Troubleshooting Voice over Frame Relay Interfaces to the IP Network 415

    Verifying the Voice Connections 415

    Verifying the Frame Relay Configuration 416

    Troubleshooting Tasks 416

    Monitoring and Maintaining the VoFR Configuration 417

    Troubleshooting VoFR with QoS 417LLQ and IP RTP Priority Commands 417Fragmentation Commands 418Frame Relay Interface Commands 418

    Troubleshooting VoIP over Frame Relay with Multipoint PVCs and Prioritization 418

    VoFR Testing Commands 419

  • Contents

    Troubleshooting Voice over ATM Interfaces to the IP Network 421

    Checking the ATM Configuration 422

    Verifying the Voice Connection 422

    Verifying the ATM Interface Configuration 423

    Verifying the VoATM Connection 425Troubleshooting Tips 425

    P A R T 4 Troubleshooting Cisco IOS Telephony Applications

    Troubleshooting Voice Applications 429

    Troubleshooting Tcl IVR 429xvCisco IOS Voice Troubleshooting and Monitoring Guide

    Testing and Debugging Your Script 430Loading Your Script 431Associating Your Script with an Inbound Dial Peer 432Displaying Information About IVR Scripts 432Using URLs in IVR Scripts 435

    URLs for Loading the IVR Script 435URLs for Loading Audio Files 436

    Tips for Using Your Tcl IVR Script 436

    Media Inactive Call Detection 436Prerequisites for Media Inactive Call Detection 437Restrictions for Media Inactive Call Detection 437Information about Media Inactive Call Detection 437

    Improved Functionality of Media Inactive Call Detection 437Modifications to Information Tags and Internal Error Codes 438

    Configuring Media Inactive Call Detection 442Output Examples for Media Inactive Call Detection 443

    show call active voice brief Command: Example 443show running config Command: Example 445Sample Tcl IVR script 449

    Troubleshooting Cisco VoiceXML 452Debugging Cisco VoiceXML Applications 453Error Events 454JavaScript or ECMA Script 455Troubleshooting Speech Recognition and Synthesis 455Troubleshooting ASR and TTS Server Functionality 456

    MGCP Scripting Overview 458

    Events and Status Codes 458Events 459

  • Contents

    xviCisco IO

    Status Codes 461Authentication Status 462Authorization Status 462Digit Collection Status 462Consult Response 463Consult Status 463Disconnect Cause 463Facility 465Feature Type 466Leg Setup Status 466Media Status 467S Voice Troubleshooting and Monitoring Guide

    Transfer Status 467VoiceXML Dialog Completion Status 468

    Troubleshooting AAA and Billing Applications 469

    Troubleshooting AAA for Voice 469Using Debug Commands for AAA Voice Troubleshooting 469

    debug radius 469debug radius accounting 477

    Using show Commands for AAA Voice Troubleshooting 480show call accounting voice summary 480show call accounting-template voice 480show call aaa attributes 481

    Accounting Server Connectivity Failure and Recovery Detection 483Prerequisites for Accounting Server Connectivity Failure and Recovery Detection 483Restrictions for Accounting Server Connectivity Failure and Recovery Detection 483Information About Accounting Server Connectivity Failure and Recovery Detection 484

    Global Accounting Script 484How to Configure Accounting Server Connectivity Failure and Recovery Detection 484

    Configuring the GAS 485Loading the GAS 486Starting the GAS 487Verifying the GAS 488Troubleshooting Accounting Server Connectivity Failure and Recovery Detection 489

    Configuration Examples for Accounting Server Connectivity Failure and Recovery Detection 491Configuring the GAS: Example 491Loading the GAS: Example 493Starting the GAS: Example 493Verifying the GAS: Example 493

    Troubleshooting Enhanced Billing Support for SIP Gateways 496

  • Contents

    Troubleshooting Settlement 497Settlement Database Not Set Up Properly 497Tcl IVR Script Not Called 497No Destination Pattern Set 498No Session Target Settlement Set on Originating Gateway 498No VoIP Inbound Dial Peer on Terminating Gateway 498No Application Attribute on Terminating Gateway 499Terminating Gateway Not Synchronized with Settlement Server 499Settlement Provider Not Running 499Router and Server Not Using SSL to Communicate 499Multiple Dial Peers Have Random Order 500xviiCisco IOS Voice Troubleshooting and Monitoring Guide

    H.323 Setup Connection Timeout 500Problem Isolation 500

    Troubleshooting Fax Applications 503

    Fax Call Flow 503Fax Pass-Through and Fax Pass-Through with Upspeed 504Cisco Fax Relay 505

    Cisco Fax Relay Fax Setup Phase 505Cisco Fax Relay Data Transfer Phase 506

    T.38 Fax Relay 507H.323 T.38 Fax Relay 507SIP T.38 Fax Relay 508MGCP T.38 Fax Relay 509

    T.37 Store-and-Forward Fax 509

    Fax Relay 510Identifying and Isolating the Problem 511Checking Basic Connectivity 512

    Normal Voice Connectivity Problems 512Configuration Problems Related to Dial Peers 512Other Basic Connectivity Problems Not Relating to Dial Peers 515Fax Connectivity Problems Across the PSTN 515

    Checking for Slips and Other Errors on Digital Interfaces 515Checking Fax Interface Type 516Ensuring That the Fax Codec is Loaded During the Fax Call 516Disabling Fax Relay and Change Codec for Pass-Through 517Checking for Packet Loss on the Voice Network 518Disabling Fax Relay ECM (Cisco Proprietary VoIP Only) 518Enabling T.38 Packet Redundancy (T.38 VoIP Only) 519Setting the Fax NSF Command to All Zeroes 519

  • Contents

    xviiiCisco IO

    Achieving the Final Stages of Resolution 519Debugging Fax Relay 520

    T.30 Messages 520Fax Relay Debug Commands 522Fax Analyzers 524

    Fax Detection 525

    P A R T 5 Monitoring the Cisco IOS Voice Network

    Monitoring the Voice Network 529

    Periodic Monitoring Tasks 529S Voice Troubleshooting and Monitoring Guide

    Tools for Monitoring the VoIP Network 530Cisco Voice Manager 530Quality of Service Device Manager 530Cisco Service Assurance Agent 531CiscoWorks Voice Health Monitor 531Cisco Gateway Management Agent 532Cisco QoS Policy Manager 532

    Voice Performance Statistics on Cisco Gateways 533

    Prerequisites for Voice Performance Statistics on Cisco Gateways 533

    Restrictions for Voice Performance Statistics on Cisco Gateways 533

    Information About Voice Performance Statistics on Cisco Gateways 534Basic Terminology and Feature Design 534

    What Are the Types of Accounting Statistics? 535What Are the Types of Signaling Statistics and Aggregation Levels? 535What Are IECs? 536

    Management of the Statistical Collection 537What Are the Allowable Time Ranges? 537What Are Thresholds? 537What Are the Allowable Storage Capacities? 538How Is Memory Used? 538

    Management of the Archive Process 538Display of Records and Time Ranges 539

    What Records Are Displayed Since System Reset or Reboot? 539What Time Ranges Are Displayed? 539

    Voice Interface Changes During Call-Statistics Collection Periods 540Addition or Removal of a Voice Port 540Configuration Change of Any Trunk Group 540

    Benefits of Voice Performance Statistics on Cisco Gateways 541

  • Contents

    Configuring Voice Performance Statistics on Cisco Gateways 541Summary of Configuration Tasks 542Configuring the Duration and Time Periods of Call Statistics on the Gateway 543

    Configuring the Gateway to Collect Call Statistics on a Periodic Basis 543Configuring the Gateway to Collect Call Statistics Since the Last Reset 544Configuring the Gateway to Collect Call Statistics for a Specific Time Interval 545

    Configuring the Gateway to Collect Signaling Statistics 546Enabling the Gateway to Collect Signaling Statistics 546Configuring the Minimum Call Duration and Signaling Thresholds 548Disabling the Collection of Signaling Statistics 549Displaying the Signaling Statistics for Each Aggregation Level 550xixCisco IOS Voice Troubleshooting and Monitoring Guide

    Clearing Signaling Statistics 556Configuring the Gateway to Collect VoIP AAA Accounting Statistics 557

    Prerequisites 557Restrictions 557Enabling the Collection of VoIP AAA Accounting Statistics on the Gateway 557Configuring a Designated Server Group for a Broadcast Method List 559Disabling the Collection of VoIP AAA Accounting Statistics 560Displaying the VoIP AAA Accounting Statistics 561Clearing the VoIP AAA Accounting Statistics 562Troubleshooting Tips for VoIP AAA Accounting Statistics 563

    Managing the Collection of Voice Statistics 566Configuring the FTP Server to Enable Archiving of Statistics from the Gateway 566Configuring the Gateway to Archive Statistics to an FTP Server 569Configuring the Gateway to Archive Statistics to a Syslog Server 570Displaying Memory Usage 571Displaying All Statistics and Pushing Them to an FTP or Syslog Server 572Clearing the Collected Call Statistics 572Monitoring the Statistical Reporting 573

    Configuration Examples for Voice Performance Statistics on Cisco Gateways 581User-Specific Configurations for Call Statistic Collection: Example 582Manually Clearing Statistics: Example 582Collection of Aggregation-Level Statistics: Example 583Memory Usage: Example 584Collection of Statistics Since System Reset: Examples 584Designated Server Group: Example 587Location of the FTP Server: Example 587Maximum File Size for the Syslog Server: Example 587Maximum Duration for Storage: Example 587

  • Contents

    xxCisco IO

    Cause Codes and Debug Values 589

    Details of Cause Codes and Debug Values for VoIP 589Q.931 Call Disconnection Causes 589Codec Negotiation Values 590Tone Types 591FAX-Rate and VAD Capabilities Values 591

    Internal Cause Codes for SIP and H.323 592

    IN D E XS Voice Troubleshooting and Monitoring Guide

  • About Cisco IOS Software Documentation

    This chapter discusses the objectives, audience, organization, and conventions of Cisco IOS Voice Troubleshooting and Monitoring Guide documentation. It also provides sources for obtaining documentation from Cisco Systems.

    Document

    Audience

    DocumentxxiCisco IOS Voice Troubleshooting and Monitoring Guide

    ation ObjectivesThis document provides information about troubleshooting and maintain Cisco voice networking devices. Links are provided for additional troubleshooting information.

    The Cisco IOS software documentation set is intended primarily for users who configure and maintain Cisco networking devices (such as routers and switches) but who may not be familiar with the tasks, the relationship between tasks, or the Cisco IOS software commands necessary to perform particular tasks.

    ation OrganizationThis document includes the following sections:

    Guide to Cisco IOS Voice Troubleshooting and Monitoring Features Troubleshooting Cisco IOS Voice Overview

    Voice Call Flow Overview Debug Command Output on Cisco IOS Voice Gateways Filtering Troubleshooting Output Cisco VoIP Internal Error Codes Troubleshooting Resources

    Troubleshooting Cisco IOS Voice Telephony Troubleshooting Analog Voice Interfaces to the IP Network Troubleshooting Digital Voice Interfaces to the IP Network Troubleshooting Quality of Service for VoIP

  • About Cisco IOS Software DocumentationDocumentation Organization

    xxiiCisco

    Troubleshooting Cisco IOS Voice Technologies Troubleshooting H.323 Interfaces to the IP Network Troubleshooting SIP Interfaces to the IP Network Troubleshooting MGCP and Related Protocol Interfaces to the IP Network Troubleshooting Voice over Frame Relay Interfaces to the IP Network Troubleshooting Voice over ATM Interfaces to the IP Network

    Troubleshooting Cisco IOS Telephony Applications Troubleshooting Voice Applications Troubleshooting AAA and Billing Applications

    Troubleshooting Fax Applications Monitoring the Cisco IOS Voice Network

    Monitoring the Voice Network

    Supporting D

    Document or Resou

    Cisco IOS Voice CoCisco IOS Voice Co

    Cisco IOS Debug C

    Cisco IOS ConfiguManagement Comm

    Internetwork TroubCisco IOS Software

    Cisco 2600 series r

    Cisco 3600 series r

    Cisco 3700 series r IOS Voice Troubleshooting and Monitoring Guide

    Voice Performance Statistics on Cisco Gateways Cause Codes and Debug Values

    ocuments and Resources

    rce Description

    nfiguration Library Explains the configuration of voice features on Cisco IOS platforms.mmand Reference Explains the function and syntax of the Cisco IOS voice software

    commands. For more information about defaults and usage guidelines, refer to the Cisco IOS command reference publications.

    ommand Reference Contains an alphabetical listing of the debug commands and their descriptions. Documentation for each command includes a brief description of its use, command syntax, usage guidelines, and sample output.

    ration Fundamentals and Network and Reference

    Contains an alphabetical listing of commonly used Cisco IOS commands and their descriptions. Documentation for each command includes a brief description of its use, command syntax, usage guidelines, and sample output.

    leshooting Handbook Contains general troubleshooting information for Cisco platforms. System Messages Lists and describes Cisco IOS system messages. Not all system

    messages indicate problems with your system. Some are informational, and others may help diagnose problems with communications lines, internal hardware, or the system software.

    outer product documentation Contains installation and configuration information for Cisco 2600 series routers, network modules, and interface cards.

    outer product documentation Contains installation and configuration information for Cisco 3600 series routers, network modules, and interface cards.

    outer product documentation Contains installation and configuration information for Cisco 3700 series routers, network modules, and interface cards.

  • About Cisco IOS Software DocumentationDocumentation Organization

    Cisco AS5350 product documentation Contains installation and configuration information for Cisco AS5350 universal gateways and cards.

    Cisco AS5400 product documentation Contains installation and configuration information for Cisco AS5400 universal gateways and cards.

    Cisco AS5800 product documentation Contains installation and configuration information for Cisco AS5800 access servers and cards.

    Cisco AS5850 product documentation Contains installation and configuration information for Cisco AS5850 universal gateways and cards.

    MIX-Multichannel T1/E1 Port Adapter Installation and Configuration Guide

    Describes the installation and technical aspects of Cisco MIX-Multichannel T1/E1 port adapter hardware.

    Dictionary of Internetworking Terms and Acronyms Compiles and defines the terms and acronyms used in the internetworking industry.

    New Cisco IOS feature documentation Documents the mainline release of Cisco IOS software (for example,

    Release notes

    RFCs

    MIBs

    Document or Resource DescriptionxxiiiCisco IOS Voice Troubleshooting and Monitoring Guide

    Cisco IOS Release 12.3). New software features are introduced in early deployment releases (for example, the Cisco IOS T release train for 12.3, 12.3(x)T). Documentation for these new features can be found in standalone documents called feature modules. Feature module documentation describes new Cisco IOS software and hardware networking functionality and is available on Cisco.com.Describes system requirements, provides information about new and changed features, and includes other useful information about specific software releases.RFCs are standards documents maintained by the Internet Engineering Task Force (IETF). Cisco IOS software documentation references supported RFCs when applicable. The full text of referenced RFCs may be obtained at http://www.rfc-editor.org/.MIBs are used for network monitoring. For lists of supported MIBs arranged by platform and release, and to download MIB files, see the Cisco MIB website on Cisco.com at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.

  • About Cisco IOS Software DocumentationDocument Conventions

    xxivCisco

    Document ConventionsWithin Cisco IOS software documentation, the term router is generally used to refer to a variety of Cisco products (for example, routers, access servers, and switches). Routers, access servers, and other networking devices that support Cisco IOS software are shown interchangeably within examples. These products are used only for illustrative purposes; that is, an example that shows one product does not necessarily indicate that other products are not supported.The Cisco IOS documentation set uses the following conventions:

    Convention Description

    ^ or Ctrl The ^ and Ctrl symbols represent the Control key. For example, the key combination ^D or Ctrl-D means hold down the Control key while you press the D key. Keys are indicated in capital letters but are not case sensitive.

    string A string is a nonquoted set of characters shown in italics. For example, when setting an SNMP

    Convention

    boldfaceitalics[x]|[x | y]

    {x | y}

    Convention

    [x {y | z}]

    Conventionscreen

    boldface screen

    < > IOS Voice Troubleshooting and Monitoring Guide

    Command syntax descriptions use the following conventions:

    Nested sets of square brackets or braces indicate optional or required choices within optional or required elements. For example:

    Examples use the following conventions:

    community string to public, do not use quotation marks around the string or the string will include the quotation marks.

    Description

    Boldface text indicates commands and keywords that you enter literally as shown.Italic text indicates arguments for which you supply values.Square brackets enclose an optional element (keyword or argument).A vertical line indicates a choice within an optional or required set of keywords or arguments.Square brackets enclosing keywords or arguments separated by a vertical line indicate an optional choice.Braces enclosing keywords or arguments separated by a vertical line indicate a required choice.

    Description

    Braces and a vertical line within square brackets indicate a required choice within an optional element.

    Description

    Examples of information displayed on the screen are set in Courier font.Examples of text that you must enter are set in Courier bold font.Angle brackets enclose text that is not printed to the screen, such as passwords.

  • About Cisco IOS Software DocumentationObtaining Documentation

    The following conventions are used to attract the attention of the reader:

    Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the manual.

    Timesaver Means the described action saves time. You can save time by performing the action described in the paragraph.

    Tip

    Caution

    Warning

    Obtaining

    Cisco.com

    ! An exclamation point at the beginning of a line indicates a comment line. (Exclamation points are also displayed by the Cisco IOS software for certain processes.)

    [ ] Square brackets enclose default responses to system prompts.

    Convention DescriptionxxvCisco IOS Voice Troubleshooting and Monitoring Guide

    Means the following information will help you solve a problem. The tips information might not be troubleshooting or even an action, but could be useful information, similar to a Timesaver.

    Means reader be careful. In this situation, you might do something that could result in equipment damage or loss of data.

    IMPORTANT SAFETY INSTRUCTIONS

    This warning symbol means danger. You are in a situation that could cause bodily injury. Before you work on any equipment, be aware of the hazards involved with electrical circuitry and be familiar with standard practices for preventing accidents. Use the statement number provided at the end of each warning to locate its translation in the translated safety warnings that accompanied this device.

    DocumentationCisco documentation and additional literature are available on Cisco.com. Cisco also provides several ways to obtain technical assistance and other technical resources. These sections explain how to obtain technical information from Cisco Systems.

    You can access the most current Cisco documentation at this URL:http://www.cisco.com/univercd/home/home.htmYou can access the Cisco website at this URL:http://www.cisco.com

  • About Cisco IOS Software DocumentationDocumentation Feedback

    xxviCisco

    You can access international Cisco websites at this URL:http://www.cisco.com/public/countries_languages.shtml

    Ordering DocumentationYou can find instructions for ordering documentation at this URL:http://www.cisco.com/univercd/cc/td/doc/es_inpck/pdi.htmYou can order Cisco documentation in these ways:

    Registered Cisco.com users (Cisco direct customers) can order Cisco product documentation from the Ordering tool:http://www.cisco.com/en/US/partner/ordering/index.shtml

    Nonregistered Cisco.com users can order documentation through a local account representative by calling Cisco Systems Corporate Headquarters (California, USA) at 408 526-7208 or, elsewhere in

    Document

    Obtaining IOS Voice Troubleshooting and Monitoring Guide

    North America, by calling 800 553-NETS (6387).

    ation FeedbackYou can send comments about technical documentation to [email protected] can submit comments by using the response card (if present) behind the front cover of your document or by writing to the following address:Cisco SystemsAttn: Customer Document Ordering170 West Tasman DriveSan Jose, CA 95134-9883We appreciate your comments.

    Technical AssistanceFor all customers, partners, resellers, and distributors who hold valid Cisco service contracts, Cisco Technical Support provides 24-hour-a-day, award-winning technical assistance. The Cisco Technical Support Website on Cisco.com features extensive online support resources. In addition, Cisco Technical Assistance Center (TAC) engineers provide telephone support. If you do not hold a valid Cisco service contract, contact your reseller.

  • About Cisco IOS Software DocumentationObtaining Technical Assistance

    Cisco Technical Support WebsiteThe Cisco Technical Support Website provides online documents and tools for troubleshooting and resolving technical issues with Cisco products and technologies. The website is available 24 hours a day, 365 days a year at this URL:http://www.cisco.com/techsupportAccess to all tools on the Cisco Technical Support Website requires a Cisco.com user ID and password. If you have a valid service contract but do not have a user ID or password, you can register at this URL:http://tools.cisco.com/RPF/register/register.do

    Submitting a Service RequestUsing the online TAC Service Request Tool is the fastest way to open S3 and S4 service requests. (S3 and S4 service requests are those in which your network is minimally impaired or for which you require

    Definitions oxxviiCisco IOS Voice Troubleshooting and Monitoring Guide

    product information.) After you describe your situation, the TAC Service Request Tool automatically provides recommended solutions. If your issue is not resolved using the recommended resources, your service request will be assigned to a Cisco TAC engineer. The TAC Service Request Tool is located at this URL:

    http://www.cisco.com/techsupport/servicerequestFor S1 or S2 service requests or if you do not have Internet access, contact the Cisco TAC by telephone. (S1 or S2 service requests are those in which your production network is down or severely degraded.) Cisco TAC engineers are assigned immediately to S1 and S2 service requests to help keep your business operations running smoothly.To open a service request by telephone, use one of the following numbers:Asia-Pacific: +61 2 8446 7411 (Australia: 1 800 805 227)EMEA: +32 2 704 55 55USA: 1 800 553 2447For a complete list of Cisco TAC contacts, go to this URL:http://www.cisco.com/techsupport/contacts

    f Service Request SeverityTo ensure that all service requests are reported in a standard format, Cisco has established severity definitions.Severity 1 (S1)Your network is down, or there is a critical impact to your business operations. You and Cisco will commit all necessary resources around the clock to resolve the situation. Severity 2 (S2)Operation of an existing network is severely degraded, or significant aspects of your business operation are negatively affected by inadequate performance of Cisco products. You and Cisco will commit full-time resources during normal business hours to resolve the situation.Severity 3 (S3)Operational performance of your network is impaired, but most business operations remain functional. You and Cisco will commit resources during normal business hours to restore service to satisfactory levels.

    Severity 4 (S4)You require information or assistance with Cisco product capabilities, installation, or configuration. There is little or no effect on your business operations.

  • About Cisco IOS Software DocumentationObtaining Additional Publications and Information

    xxviiiCisco

    Obtaining Additional Publications and InformationInformation about Cisco products, technologies, and network solutions is available from various online and printed sources.

    Cisco Marketplace provides a variety of Cisco books, reference guides, and logo merchandise. Visit Cisco Marketplace, the company store, at this URL:http://www.cisco.com/go/marketplace/

    The Cisco Product Catalog describes the networking products offered by Cisco Systems, as well as ordering and customer support services. Access the Cisco Product Catalog at this URL:http://cisco.com/univercd/cc/td/doc/pcat/

    Cisco Press publishes a wide range of general networking, training and certification titles. Both new and experienced users will benefit from these publications. For current Cisco Press titles and other information, go to Cisco Press at this URL: IOS Voice Troubleshooting and Monitoring Guide

    http://www.ciscopress.com Packet magazine is the Cisco Systems technical user magazine for maximizing Internet and

    networking investments. Each quarter, Packet delivers coverage of the latest industry trends, technology breakthroughs, and Cisco products and solutions, as well as network deployment and troubleshooting tips, configuration examples, customer case studies, certification and training information, and links to scores of in-depth online resources. You can access Packet magazine at this URL:

    http://www.cisco.com/packet iQ Magazine is the quarterly publication from Cisco Systems designed to help growing companies

    learn how they can use technology to increase revenue, streamline their business, and expand services. The publication identifies the challenges facing these companies and the technologies to help solve them, using real-world case studies and business strategies to help readers make sound technology investment decisions. You can access iQ Magazine at this URL:http://www.cisco.com/go/iqmagazine

    Internet Protocol Journal is a quarterly journal published by Cisco Systems for engineering professionals involved in designing, developing, and operating public and private internets and intranets. You can access the Internet Protocol Journal at this URL:http://www.cisco.com/ipj

    World-class networking training is available from Cisco. You can view current offerings at this URL:http://www.cisco.com/en/US/learning/index.html

  • Guide to Cisco IOS Voice Troubleshooting and Monitoring Features

    This chapter contains information about Cisco IOS voice troubleshooting and monitoring features and

    NotexxixCisco IOS Voice Troubleshooting and Monitoring Guide

    includes the location of the information in this document or other documents. Use Cisco Feature Navigator to see the features a Cisco IOS software release supports by platform, and to find the image that is best for you. Access Cisco Feature Navigator at http://www.cisco.com/go/fn.

    To use Cisco Feature Navigator, you must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.

    New Cisco IOS voice troubleshooting and monitoring features are introduced in the following Cisco IOS software releases:

    Cisco IOS Release 12.4(4)T, page xxx Cisco IOS Release 12.3(8)T, page xxx Cisco IOS Release 12.3(4)T, page xxxi Cisco IOS Release 12.2(11)T, page xxxi

  • Guide to Cisco IOS Voice Troubleshooting and Monitoring FeaturesCisco IOS Release 12.4(4)T

    xxxCisco

    Cisco IOS Release 12.4(4)T

    Cisco IOS

    Feature Description Location

    MGCP Call Centric Debug Enables the filtering of MGCP debug output based on selected criteria and standardizes the format of the MGCP debug header. All MGCP debug output for a single call can be identified and correlated across the various layers in IOS software. Filtering debug output reduces extraneous information, making it easier to locate the correct information and reducing the impact to platform performance.

    MGCP Call Centric Debug section on page 96

    Test Call Provides the ability for a remote station or gateway to establish a call to any destination

    Using the Test Call Feature to Verify Voice Path section on page 342

    Voice Call Debug FH.323 Gatekeepers

    Feature

    Voice TroubleshooEnhancements IOS Voice Troubleshooting and Monitoring Guide

    Release 12.3(8)T

    address from a Test Call station located at a network operations center and to audibly verify the voice path.

    iltering on Enables selected debugging traces for voice calls. This feature allows you to filter and trace voice call debug messages based on selected filtering criteria, reducing the volume of output for more efficient troubleshooting.

    Voice Call Debug Filtering on H.323 Gatekeepers section on page 69

    Description Location

    ting Adds standardized voice debug header and voice call debug filtering capabilities to additional and replacement debug commands, enables command profile debugging, and provides data dump for internal data structures.

    Debug Command Output on Cisco IOS Voice Gateways section on page 17

  • Guide to Cisco IOS Voice Troubleshooting and Monitoring FeaturesCisco IOS Release 12.3(4)T

    Cisco IOS Release 12.3(4)T

    Cisco IOS

    Feature Description Location

    Accounting Server Connectivity Failure and Recovery Detection

    Provides the option to reject new calls entering the VoIP network and tear down all existing calls upon detecting connectivity failure to the method list that is associated with RADIUS-based accounting servers.

    Accounting Server Connectivity Failure and Recovery Detection section on page 483

    Inactive Call Detection Enhances Cisco IOS behavior for disconnecting a call when an inactive condition is detected. This feature provides more control for managing these calls.

    Media Inactive Call Detection section on page 436

    SIP Debug Output Filtering Support

    Provides the capability for SIP-related debug SIP Debug Output Filtering Support

    Voice DSP Crash D

    Voice PerformanceCisco Gateways

    VoIP Debug FilteriVoice Gateways

    VoIP Internal Error

    Feature

    Enhanced Debug CCisco Voice GatewxxxiCisco IOS Voice Troubleshooting and Monitoring Guide

    Release 12.2(11)T

    output to be filtered based on a set of user-defined matching conditions.

    section on page 83

    ump Analysis Allows Cisco IOS voice platforms using TI DSPs the ability to capture the contents of the DSP memory into a dump file if there is a DSP crash.

    Voice DSP Crash Dump File Analysis section on page 239

    Statistics on Enables the collection of voice call statistics based on user-configured time ranges. The statistics that can be collected are from the following functional areas:

    RADIUS accounting Cisco IOS generated internal error codes

    (IECs) Gateway port (interface) statistics

    Voice Performance Statistics on Cisco Gateways section on page 533

    ng on Cisco Allows you to filter and trace voice call debug messages based on selected filtering criteria, reducing the volume of output for more efficient troubleshooting.

    Voice Call Debug Filtering on Cisco Voice Gateways section on page 50

    Codes Generates IECs for gateway-detected errors that cause the gateway to release or refuse a call. IECs enhance troubleshooting for VoIP networks by helping to determine the source and reason for call termination.

    Cisco VoIP Internal Error Codes section on page 109

    Description Location

    apabilities for ays

    Enables you to identify and track a specific call in a multiple-call environment.

    Debug Command Output on Cisco IOS Voice Gateways section on page 17

  • Guide to Cisco IOS Voice Troubleshooting and Monitoring FeaturesCisco IOS Release 12.2(11)T

    xxxiiCisco IOS Voice Troubleshooting and Monitoring Guide

  • P A R T 1

    Troubleshooting Cisco IOS Voice Overview

  • Voice Call Flow Overview

    To troubleshoot problems with voice networks, you must follow the call both inside the router and outside on the network in order to isolate the problem. You must understand the relationship of dial peers and call legs to follow the calls. For detailed information on dial peers and call legs, refer to Dial Peer

    Call Setup3Cisco IOS Voice Troubleshooting and Monitoring Guide

    Configuration on Voice Gateway Routers.The following sections contain information about call flow:

    Call Setup, page 3 Call Flow Through Router Components, page 14

    For information about voice network design, refer to Troubleshooting and Debugging VoIP Call Basics, document ID 14081.

    A voice call over a packet network is segmented into discrete call legs. Each call leg is associated with a dial peer. A call leg is a logical connection between two voice gateways or between a gateway and an IP telephony device (for example, Cisco CallManager or a session initiation protocol (SIP) server). An example of POTS and VoIP call legs is shown in Figure 1.

    Figure 1 Dial Peer Call Legs

    Details for the call setup are described in the following sections: Call Setup Elements, page 4 Call Setup Process, page 6 Dial Peer Matching, page 7

    Source Destination

    Call leg 1 (POTS dial peer)

    Call leg 2 (VoIP dial peer)

    Call leg 3 (VoIP dial peer)

    Call leg 4 (POTS dial peer)

    3595

    0

    IP networkV V

  • Voice Call Flow OverviewCall Setup

    4Cisco

    Troubleshooting and debugging should focus first on each leg independently and then on the VoIP call as a whole.

    Call Setup ElementsYou can isolate where a problem is occurring by determining which dial peer or call leg is having the problem, as described in the following sections:

    Source and Destination POTS Dial Peers, page 4 Voice Network Dial Peers, page 4 Inbound and Outbound Call Legs, page 5

    Source and Destination POTS Dial Peers

    Voice Network IOS Voice Troubleshooting and Monitoring Guide

    POTS dial peers define the characteristics of a traditional telephony network connection. The POTS dial peer maps a dial string to a specific voice port on the local gateway. Normally the voice port connects the gateway to the local public switched telephone network (PSTN), PBX, or analog telephone.

    Dial Peers

    Voice network dial peers define the attributes of a packet voice network connection. Voice network dial peers map a dial string to a remote network device. Some examples of these remote network devices are as follows:

    Destination gateway Cisco CallManager SIP server Open Settlements Protocol (OSP) server (for VoIP using settlement) H.323 gatekeeper Mail Transfer Agent (MTA) server (for Multimedia Mail over IP scenarios)

    The specific type of voice-network dial peer used depends on the packet network technology. Different technologies supported on voice dial peers are as follows:

    Voice over IP (VoIP)The dial peer is mapped to the IP address, Domain Name System (DNS) name, or server-type of the destination VoIP device that terminates the call. This mapping applies to all VoIP protocols such as H.323, SIP, and Media Gateway Control Protocol (MGCP).

    Voice over Frame Relay (VoFR)The dial peer is mapped to the data-link connection identifier (DLCI) of the interface from which the call exits the router.

    Voice over ATM (VoATM)The dial peer is mapped to the ATM virtual circuit for the interface from which the call exits the router.

    Multimedia Mail over IP (MMoIP)The dial peer is mapped to the e-mail address of the Simple Mail Transfer Protocol (SMTP) server. This type of dial peer is used for store-and-forward fax (on-ramp and off-ramp faxing).

  • Voice Call Flow OverviewCall Setup

    The voice-network dial peer also sets the attributes of the network connection, such as which codec to use, the capability to do voice activity detection (VAD), and dual-tone multifrequency (DTMF) relay configuration. A voice network dial peer can point to any device that is compatible with skinny protocol (SCCP), H.323, or SIP. MGCP configuration on a Cisco IOS gateway does not use voice network dial peers.

    Voice and Billing Application Call Leg

    An IP call leg can go to an external application server. An example of a Cisco IOS VoiceXML application is shown in Figure 2. This call leg is used for voice and billing applications, such as interactive voice response (IVR).

    Figure 2 Cisco IOS VoiceXML Application Components

    Inbound and Ou

    HTTP server5Cisco IOS Voice Troubleshooting and Monitoring Guide

    tbound Call Legs

    Each call leg can be either an inbound or outbound call leg, depending on the calls direction. For a call from an IP phone to the PSTN, the VoIP call leg is the inbound call leg, and the POTS leg is the outbound call leg. The opposite is true for a call from the PSTN to the IP phone. Every call has an associated inbound and outbound call leg.

    TFTP server

    SMTP server

    RTSP server

    VoiceXML-enabled gateway

    Voice gateway 3788

    8

    IP

    IPnetworkV

    V PSTN

    PSTN

  • Voice Call Flow OverviewCall Setup

    6Cisco

    Call Setup ProcessA call is segmented into call legs with a dial peer associated with each call leg. See Figure 1.The process has eight main steps:1. The POTS call arrives at the originating gateway, and an inbound POTS dial peer is matched.

    Note At this stage, if configured on the inbound POTS dial-peer, nondefault inbound POTS services or Tool Command Language (Tcl) applications are used. When you use such services or applications, be certain that the correct inbound POTS dial peer is matched. Examples of services or applications include:

    Direct inward dialing (DID) Tcl-based applications such as IVR, VoIP SIP transfer, or on-ramp faxing for store-and-forward fax. IOS Voice Troubleshooting and Monitoring Guide

    2. After the incoming call is associated with an inbound POTS dial peer, the originating gateway creates an inbound POTS call leg and assigns it a CallEntry ID and global unique identifier (GUID). For more information about these identifiers, see the Debug Header Format section on page 18.

    3. The originating gateway uses the dialed string to match an outbound voice-network dial peer.4. After the dialed string are associated with an outbound voice network dial peer, the originating

    gateway creates an outbound voice-network call leg and assigns it a call ID for the second call leg. The preceding process is illustrated in Figure 3.

    Figure 3 Call Legs from the Perspective of the Originating Router

    5. The voice network call request arrives at the terminating gateway and an inbound voice network dial peer is matched.

    6. After the terminating gateway associates the incoming call to an inbound voice-network dial peer, the terminating gateway creates the inbound voice-network call leg and assigns it a call ID for the third call leg.At this point, both gateways negotiate voice-network capabilities and applications (if required). Default capabilities are not displayed in the gateway Cisco IOS configuration output. Use the show dial-peer voice command to display the configured capabilities, services, and applications on POTS and voice-network dial peers: Default capabilities include codec g729r8, vad enable, dtmf-relay disable, fax-relay disable,

    req-qos best-effort, acc-qos best-effort, and session protocol cisco (for H.323). Examples of Tcl applications include remote IP authentication and off-ramp faxing.

    InboundPOTS call leg

    Source Destination

    OutboundVoIP call leg 35

    946

    IP networkVV

  • Voice Call Flow OverviewCall Setup

    When nondefault capabilities or applications are requested by the originating gateway, the terminating gateways needs to match an inbound voice network dial peer that is configured for these capabilities or applications.

    7. The terminating gateway uses the dialed string to match an outbound POTS dial peer.8. After associating the incoming call setup to an outbound POTS dial peer, the terminating

    gatewaycreates an outbound POTS call leg, assigns it a call ID, and terminates the call. The preceding steps are illustrated in Figure 4.

    Figure 4 Call Legs from the Perspective of the Terminating Router

    Dial Peer M

    Source Destination

    IP networkVV7Cisco IOS Voice Troubleshooting and Monitoring Guide

    In scenarios where Cisco CallManager is present with a Cisco IOS gateway, the following assumptions apply:

    For inbound calls to Cisco CallManager through a Cisco IOS gateway, the gateway behaves as an originating device. (See Steps 1 to 4.)

    For outbound calls from Cisco CallManager through a Cisco IOS gateway, the gateway behaves as a terminating device. (See Steps 5 to 8.)

    atchingIn order for a call to be routed, the attributes set in the dial peers must be matched. Each dial peer can contain a variety of parameters. Some parameters are valid only for voice network or POTS dial peers, and others apply to all dial peers. This section covers the following topics:

    Inbound Dial Peer Matching, page 8 Outbound Dial Peer Matching, page 11 Voice Network Dial Peer Matching, page 13 POTS Dial Peer Matching, page 13

    The destination-pattern parameter on the dial peer command controls call routing. For inbound dial peers, the destination-pattern parameter is matched against the calling number, or automatic number identifier (ANI) string. For outbound dial peers, the destination-pattern parameter is matched against the called number, or dialed number identification service (DNIS) string. A dial peer with the destination-pattern parameter works for both outbound and inbound matching. Multiple dial peers can match a specific digit string. The gateway usually attempts to perform longest-match routing. For example, if you have two patterns, 555.... and 55501.., and the called party number is 5550123, the gateway matches the peer with 55501..; however, if the call to that peer fails, the gateway attempts to use the other matching peer. If more than one peer has the same destination pattern configured, the preference parameter on the dial peer can resolve the priority. The peer with the lowest preference number has the highest priority.

    InboundVoIP call leg

    OutboundPOTS call leg 36

    849

  • Voice Call Flow OverviewCall Setup

    8Cisco

    The operational status for a dial peer must be administratively up and valid for it to be matched. To be considered operational, dial peers must meet one of the following conditions:

    destination-pattern is configured and a voice-port or session target is also configured. incoming called-number is configured. answer-address is configured.

    Matching the called party number with the destination-pattern parameter is always used to match an outbound dial peer. The inbound dial peer does not affect where the call is routed, but it does determine all of the call properties for the voice network side of the call, regardless of where the outbound peer terminates.The incoming called-number and answer-address commands are used only to match inbound dial peers. They are not used for call routing or choosing an outbound dial peer. The incoming called-number command matches based on the called party number but does not play a role in where the call is routed. It is used only to select the inbound dial peer. If this match is unsuccessful, the answer-address command tries for a match using the calling party number information rather than the called party number. If both of these matches fail, the calling party information is matched against the destination-pattern command configured on the dial peers. On POTS ports, the inbound dial peer can

    Inbound Dial Pe IOS Voice Troubleshooting and Monitoring Guide

    be matched based on port configuration. If no match can be made, the dial peer 0 attribute is used. See the Dial Peer 0 section on page 10 for more information about dial peer 0.For more information about the operational status of dial peers, refer to Understanding the Operational Status of Dial Peers on Cisco IOS Platforms, document ID 12426.For more detailed information on dial peer matching, configuration steps, and information about parameters, refer to Dial Peer Features and Configuration in the Dial Peer Configuration on Voice Gateway Routers document.For additional information about dial peer matching, refer to Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms, document ID 14074.

    er Matching

    The inbound dial peer determines all of the call properties for the voice network side of the call. Inbound POTS dial peers are associated to incoming POTS call legs on the originating gateways. Inbound voice-network dial peers are associated to incoming voice-network call legs of the terminating gateway.To match inbound call legs to dial peers, the router uses three information elements in the call setup message and five configurable dial peer attributes. The three call setup elements are:

    Called number or DNISA set of numbers representing the destination, which is derived from the ISDN setup message or channel associated-signaling (CAS) DNIS.

    Calling number or ANIA set of numbers representing the origin, which is derived from the ISDN setup message or CAS ANI.

    Voice portThe voice port carrying the call.

    The five configurable dial peer attributes are: Incoming called numberA string representing the called number or DNIS. It is configured by

    using the incoming called-number dial-peer voice configuration command in POTS or MMoIP dial peers.

    Answer addressA string representing the calling number or ANI. It is configured by using the answer-address dial-peer voice configuration command in POTS or VoIP dial peers and is used only for inbound calls from the IP network.

    Destination patternA string representing the calling number or ANI. It is configured by using the destination-pattern dial-peer voice configuration command in POTS or voice-network dial peers.

  • Voice Call Flow OverviewCall Setup

    ApplicationA string representing the predefined application that you enable on the dial peer. It is configured by using the application dial-peer voice configuration command on inbound POTS dial peers.

    PortThe voice port through which calls to this dial peer are placed.When the gateway receives a call setup request, a dial-peer match is made for the incoming call to facilitate routing the call to different session applications. The gateway does not perform digit-by-digit matching. Instead, the full digit string received in the setup request is used for matching against configured dial peers.The gateway selects an inbound dial peer by matching the information elements in the setup message with the dial peer attributes. The gateway matches these items in the following order:1. Called number or DNIS with incoming called-number

    First, the gateway attempts to match the called number of the call setup request with the configured incoming called-number parameter of each dial peer. Because call setups always include DNIS information, Cisco recommends using the incoming called-number command for inbound dial peer

    Note9Cisco IOS Voice Troubleshooting and Monitoring Guide

    matching. This attribute has matching priority over the answer-address and destination-pattern parameter.

    2. Calling number or ANI with answer-addressIf no match is found in Step 1, the gateway attempts to match the calling number of the call setup request with the answer-address of each dial peer. This attribute may be useful in situations where you want to match calls based on the calling number (originating).

    3. Calling number (ANI) with destination-patternIf no match is found in step 2, the gateway attempts to match the calling number of the call setup request to the destination-pattern of each dial peer.

    4. Voice port (associated with the incoming call setup request) with the configured dial peer port parameter (applicable for inbound POTS call legs).If no match is found in Step 3, the gateway attempts to match the configured dial-peer port parameter to the voice port associated with the incoming call. If multiple dial peers have the same port configured, the dial peer first added in the configuration is matched.

    If no match is found using these steps, then dial peer 0 is used. See the Dial Peer 0 section on page 10 for more information about dial peer 0.

    Step 4 is not applicable to voice or dial platforms such as the Cisco AS5300 access server, Cisco AS5350 universal gateway, Cisco AS5400 universal gateway, Cisco AS5800 universal gateway, and Cisco AS5850 universal gateway. If any one of first three steps are not used, then dial-peer 0 is matched and the call is treated as a dial modem call. This call treatment can result in getting modem tones as opposed to dial tone for inbound calls.

    Only one condition must be met for the gateway to select a dial peer. The gateway stops searching as soon as one dial peer is matched. It is not necessary for all the attributes to be configured in the dial peer or that every attribute match the call setup information. The longest prefix matching criteria applies while each step is performed. At each step, if multiple matches are found, the one with the longest explicit match is chosen.For more information on dial-peer matching, configuration steps, and information about parameters, refer to Dial Peer Features and Configuration in the Dial Peer Configuration on Voice Gateway Routers document.

  • Voice Call Flow OverviewCall Setup

    10Cisco

    Dial Peer 0

    If no inbound peer can be matched by the defined criteria, the inbound peer is set to dial peer 0, which sometimes appears as pid:0. The characteristics of dial peer 0 cannot be changed.

    Note Cisco universal gateways, such as the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850, require configured inbound dial peers to match incoming POTS calls in order to be accepted as a voice call. If there is no inbound dial peer match, the call is treated and processed as a dialup (modem) call.

    For an inbound voice network call, dial peer 0 has the following characteristics: Supports any codec No DTMF relay

    IP precedence 0 VAD-enabled

    Note

    Inbound Dial Peer C IOS Voice Troubleshooting and Monitoring Guide

    No RSVP support Fax-rate voice

    Dial peer 0 fails to negotiate nondefault capabilities, services, and applications, such as DTMF relay or disabled VAD.For an incoming POTS call, dial peer 0 has the following characteristics:

    No applications

    No DID

    Avoid using dial peer 0. Having the incoming called-number parameter configured correctly ensures that the dial peer is always matched with the parameters you want when placing outbound calls through a gateway. Many problems with calling out through a Cisco IOS gateway are due to codec, VAD, and DTMF-relay misconfigurations when dial peer 0 is being matched. To display which dial peers are being matched for an active call, use the show call active voice brief command.

    onfiguration Tips

    For inbound dial peers, the following configuration tips apply to certain configurations:

    Session Target Command

    For inbound dial peers, the session target command is ignored.

    ISDN Overlap Configuration

    When the timer (T) character is included at the end of the destination pattern, the router collects dialed digits until the interdigit timer expires (10 seconds, by default) or until you dial the termination character (the default is #). The timer character must be an uppercase T. In most cases, you must configure the T indicator only when the router uses two-stage dialing. If DID is configured in the inbound POTS dial peer, the router uses one-stage dialing, which means that the full dialed string is used to match outbound dial peers. The only exception is when the isdn overlap-receiving command is configured; the ISDN overlap-receiving feature requires the T indicator. When the isdn overlap-receiving command is configured on ISDN interfaces, dial peers are checked for matches after every digit is received at the ISDN layer. If a full match is made, the call is routed immediately without waiting for additional digits. The T indicator can be used to suspend this

  • Voice Call Flow OverviewCall Setup

    digit-by-digit matching and force the gateway to wait until all digits are received. The T refers to the T302 interdigit timer at the ISDN level, configurable under the serial interface associated with the ISDN interface. ISDN also provides other mechanisms to indicate the end of digits, such as setting the Sending Complete Information Element (IE) in a Q.931 information message.

    Empty Calling Field with Variable-Length Dial Plans

    In some voice network configurations, variable-length dial plans are required, especially if the network connects two or more countries where telephone number strings could be different lengths. However, this configuration can result in the calling number field being replaced with an empty number.If you enter the T character in the destination-pattern parameter for your dial peer, the router can be configured to accept a fixed-length dial string, and then wait for additional dialed digits. For example, the following dial peer configuration shows how the T character can be set to allow variable-length dial strings: dial-peer voice 1 pots destination-pattern 9T port 1/0:1

    Outbound Dial P11Cisco IOS Voice Troubleshooting and Monitoring Guide

    If an incoming call arrives with no calling number information and is matched with the POTS dial peer shown based on the destination-pattern 9T, the gateway uses the 9 digit as the calling number and forwards the call to the corresponding device. To eliminate this behavior of replacing the empty calling number field, create a dummy POTS dial peer with just the incoming called-number command configured. Because the incoming called-number statement has higher priority than the destination-pattern parameter for inbound POTS matching, dial-peer voice 2 is the POTS dial peer that is used:dial-peer voice 1 pots destination-pattern 9T port 1/0:1!dial-peer voice 2 pots incoming called-number .

    eer Matching

    The method a router uses to select an outbound dial peer depends on whether DID is configured in the inbound POTS dial peer. If DID is not configured in the inbound POTS dial peer, the router collects the incoming dialed string digit by digit. When one dial peer is matched, the router immediately places the call using the configured attributes in the matching dial peer.If DID is configured in the inbound POTS dial peer, the router uses the full incoming dial string to match the destination pattern in the outbound dial peer. With DID, the setup message contains all the digits necessary to route the call; no additional digit collection is required. If more than one dial peer matches the dial string, all of the matching dial peers are used to form a rotary group. The router attempts to place the outbound call leg using all of the dial peers in the rotary group until one is successful. For matching outbound dial peers, the gateway uses the destination-pattern dial peer command.

    On POTS dial peers, the port command is used to forward the call. On voice network dial peers, the session target command is used to forward the call.

  • Voice Call Flow OverviewCall Setup

    12Cisco

    DID Configuration

    An example of incoming dial-peer configured with DID follows:dial-peer voice 1 pots incoming called-number 81690 voice-port 0:D direct-inward-dial

    On DID calls, also referred to as one-stage dialing, the setup message contains all digits necessary to route the call, and the gateway should not do subsequent digit collection. When the gateway searches for an outbound dial peer, it uses the entire incoming dial string. This matching is by default variable-length. It is not done digit by digit because by DID definition all digits have been received. The following example helps clarify this concept:

    If the DID dial-string is 81690, the router matches dial peer 4 and forwards the complete dial-string 81690.dial-peer voice 3 voip destination-pattern 816

    Two-Stage Dialing

    Variable-Length Dia IOS Voice Troubleshooting and Monitoring Guide

    session target ipv4:172.22.10.1!dial-peer voice 4 voip destination-pattern 81690 session target ipv4:172.22.10.1

    Configuration

    If DID is not configured on the matched incoming dial peer, the gateway enters digit collection mode, called two-stage dialing. Digits are collected inband. Outbound dial peer matching is done on a digit-by-digit basis. The gateway checks for dial peer matches after receiving each digit and then routes the call when a full match is made. In this example, the dial string is 81690. Immediately after the router receives the digit 6, it matches dial peer 3 and routes the call, forwarding only the digits 816.dial-peer voice 3 voip destination-pattern 816 session target ipv4:172.22.10.1!dial-peer voice 4 voip destination-pattern 81690 session target ipv4:172.22.10.1

    In this example, dial-peer 3 is configured for wild card matching:dial-peer voice 3 voip destination-pattern 816.. session target ipv4:172.22.10.1 !dial-peer voice 4 voip destination-pattern 81690 session target ipv4:172.22.10.1

    In this case, the longest-prefix rule applies and dial peer 4 is matched for the outbound call leg.

    l Plans

    There are situations where expected dial strings do not have a set number of digits. In such cases, it is usually best to use variable-length dial peers by configuring the T terminator on the dial-peer destination-pattern command.

  • Voice Call Flow OverviewCall Setup

    The T terminator forces the gateway to wait until the full dial string is received by doing the following: Waiting for a specified interdigit timeout before routing the call. Routing the call once it receives the # termination character in the dial string. For example, if you

    dialed 5550112#, the # would indicate to the router that you dialed all the digits and that all digits before the # should be used to match a dial peer.

    In the following example, the router receives a call setup from the network with dial string 95550112. Dial peer 2 then forwards the digits 5550112 to the PSTN. dial-peer voice 2 pots destination-pattern 9T port 2/0:23

    In the following example, the dial string from an inbound POTS interface is 81690:dial-peer voice 3 voip destination-pattern 8T session target ipv4:172.22.10.1!

    Note

    Voice Network

    POTS Dial Peer13Cisco IOS Voice Troubleshooting and Monitoring Guide

    dial-peer voice 4 voip destination-pattern 81690T session target ipv4:172.22.10.1

    In this case, the longest-prefix rule applies, and dial peer 4 is matched for the outbound call leg.

    The default interdigit timeout is set for 10 seconds. To modify this value, use the timeouts interdigit voice-port command.Anytime the T is used with destination-pattern parameter, it must be preceded by a . or digits (.T or 555T, for example). Using T on its own causes the dial peers to act improperly and affects how calls are handled by the router.

    Dial Peer Matching

    When a call comes in through VoIP, the voice gateway searches for an inbound peer by looking through all the configured voice network dial peers. Using regular digit matching rules, it tries to match the peer in the following order:1. Match the called number with the incoming called number.2. Match the calling number with the answer-address parameter.3. Match the calling number with the destination pattern.4. Otherwise, use dial peer 0. See the Dial Peer 0 section on page 10 for more information.

    Matching

    When a call comes in through POTS, the voice gateway searches for an inbound peer by looking through all the configured POTS dial peers. Using regular digit-matching rules, it tries to match the peer in the following order:1. Match the called number with the incoming called number.2. Match the calling number with the answer-address parameter.3. Match the calling number with the destination pattern.

  • Voice Call Flow OverviewCall Flow Through Router Components

    14Cisco

    If the inbound interface is not PRI or BRI, or if a PRI or BRI interface does not match a dial peer using the preceding three rules, the voice gateway matches a POTS dial peer that has the inbound port configured if any of the following parameters are configured:

    destination-pattern answer-address incoming called-number

    The answer-address parameter overwrites the calls calling-party number or ANI.

    Call Flow Through Router ComponentsYou need to understand the call path through the router in order to determine where a problem is occurring. The call path through a router is shown in Figure 5. IOS Voice Troubleshooting and Monitoring Guide

    Figure 5 Router Call Flow

    The following definitions explain the function of the main components displayed in the router call flow diagram:

    Call control application programming interface (CCAPI)Three clients make use of the CCAPI: command-line interface (CLI), Simple Network Management Protocol (SNMP) agent and Session Application. The CCAPI main functions are: Identify the call legs (Which dial peer is it? Where did it come from?). Decide which session application takes the call (Who handles it?). Invoke the packet handler. Conference the call legs together. Start recording call statistics.

    Session application and dial plan mapperThe session application uses the dial plan mapper to map a number to a dial peer (local POTS or remote VoIP). The dial plan mapper uses the dial peer table to find active dial peers.

    Telephony and VoIP service provider interface (SPI)The telephony SPI communicates with the POTS dial peers. The VoIP SPI is the specific interface to the VoIP peers. Telephony or digital signal processor (DSP) drivers deliver services to the Telephony SPI; the VoIP SPI relies on session protocols.

    8897

    0

    CLI

    TelephonySPI

    SNMPSession

    applicationDial planmanner

    VoIPSPI

    Call control API

    VoFRSPI

    Calltable

    Interfacetable

    Dial peertable

  • Voice Call Flow OverviewCall Flow Through Router Components

    Telephony Interface ArchitectureFigure 6 shows the architecture of Cisco router telephony building blocks and how they interact with each other.

    Figure 6 Telephony Interface Call Flow

    Setup/release requestSetup/releaseind/response

    Applicationresponse

    Applicationrequest

    Voice processormodule

    Voice telephonyservice provider

    (VTSP)

    CCAPI15Cisco IOS Voice Troubleshooting and Monitoring Guide

    The elements in Figure 6 have the following functions and definitions: CCAPISoftware entity that establishes, terminates, and bridges call legs. Voice telephony service provider (VTSP)Cisco IOS process that services requests from the

    CCAPI and formulates the appropriate requests to the DSPs or the voice processor module (VPM). Voice processor module (VPM ) The VPM is in charge of bridging and coordinating signaling

    processes between the telephony port signaling state machine (SSM), the DSP resource manager, and the VTSP.

    DSP resource manager (DSPRM)The DSPRM provides interfaces by which the VTSP can send and receive messages to and from the DSPs.

    Packet handlerThe packet handler forwards packets between the DSPs and the peer call legs. Call peerThe call peer can be another telephony voice connection (POTS), VoFR, VoATM, or a

    VoIP connection.88

    971

    FXO SSM FXS SSM E&M SSM DSP resource manager

    DSP

  • Voice Call Flow OverviewCall Flow Through Router Components

    16Cisco

    Voice Application InterfaceFor voice applications, the router interacts with an application server. In the example in Figure 7, the call from the router is routed through the application module to the AAA server.

    Figure 7 Voice Application Interface

    Application-TCL

    Session control

    Call control API IOS Voice Troubleshooting and Monitoring Guide

    Telephony SPI

    Signaling

    Dial peer table AAA RADIUS

    TCP/UDP

    IP network

    8898

    8

  • Debug Command Output on Cisco IOS Voice Gateways

    The debugging capability for Cisco voice gateways enables you to identify and track a specific call in a

    Voice Deb17Cisco IOS Voice Troubleshooting and Monitoring Guide

    multiple-call environment. This capability allows you to correlate call information between gateways or to identify specific debug messages associated with a single call when multiple voice calls were simultaneously active.

    Voice debug output contains a standardized header to the debug outputs of multiple voice modules, such as voice telephony service provider (VTSP), call control applicat