Cisco routers provide numerous integrated commands to assist you in monitoring and troubleshooting your internetwork:

• show commands help you monitor installation behavior and normal network behavior, and isolate problem areas.

• debug commands help you isolate protocol and configuration problems.

• ping commands help you determine connectivity between devices on your network.

• trace commands provide a method of determining the route by which packets reach their destination.

This chapter discusses use of show and debug commands.

Note	Under moderate traffic loads, debug commands produce a high volume of output. We therefore recommend that, as a general rule, you use show commands first and use debug commands with caution.

Generally, you should proceed as follows:

1. Determine whether or not VoIP is working.

2. Determine whether or not you can make a voice call.

3. Verify that SIP-supported codecs are used. Support for codecs varies on different platforms; use the codec ? command to determine the codecs available on a specific platform.

4. Isolate and reproduce the failure.

5. Collect relevant information from show and debug commands, configuration files, and protocol analyzers.

6. Identify the first indication of failure in protocol traces or internal debug command output.

7. Look for the cause in configuration files.

Note	General troubleshooting of problems affecting basic functionality such as dial peers, digit translation, and IP connectivity is beyond the scope of this chapter. For links to additional troubleshooting help, see "Additional References".

Note	Commands are listed in alphabetical order.

• Use the debug aaa authentication command to display high-level diagnostics related to AAA logins.

• Use the debug asnl events command to verify that the SIP subscription server is up. The output displays a pending message if, for example, the client is unsuccessful in communicating with the server.

• Use the debug call fallback family of commands to display details of VoIP call fallback.

• Use the debug cch323 family of commands to provide debugging output for various components within an H.323 subsystem.

• Use the debug ccsip family of commands for general SIP debugging, including viewing direction-attribute settings and port and network address-translation traces. Use any of the following related commands:

  • debug ccsip all-- Enables all SIP-related debugging
  • debug ccsip calls --Enables tracing of all SIP service-provider interface (SPI) calls
  • debug ccsip error --Enables tracing of SIP SPI errors
  • debug ccsip events --Enables tracing of all SIP SPI events
  • debug ccsip info --Enables tracing of general SIP SPI information, including verification that call redirection is disabled
  • debug ccsip media --Enables tracing of SIP media streams
  • debug ccsip messages --Enables all SIP SPI message tracing, such as those that are exchanged between the SIP user-agent client (UAC) and the access server
  • debug ccsip preauth --Enables diagnostic reporting of authentication, authorization, and accounting (AAA) preauthentication for SIP calls
  • debug ccsip states --Enables tracing of all SIP SPI state tracing
  • debug ccsip transport --Enables tracing of the SIP transport handler and the TCP or User Datagram Protocol (UDP) process

• Use the debug isdn q931 command to display information about call setup and teardown of ISDN network connections (layer 3) between the local router (user side) and the network.

• Use the debug kpm l command to enable debug tracing of KPML parser and builder errors.

• Use the debug radius command to enable debug tracing of RADIUS attributes.

• Use the debug rpms-proc preauth command to enable debug tracing on the RPMS process for H.323 calls, SIP calls, or both H.323 and SIP calls.

• Use the debug rtr trace command to trace the execution of an SAA operation.

• Use the debug voip family of commands, including the following:

  • debug voip ccapi protoheaders -- Displays messages sent between the originating and terminating gateways. If no headers are being received by the terminating gateway, verify that the header-passing command is enabled on the originating gateway.
  • debug voip ivr script-- Displays any errors that might occur when the Tcl script is run
  • debug voip rtp session named-event 101 -- Displays information important to DTMF-relay debugging, if you are using codec types g726r16 or g726r24. Be sure to append the argument 101 to the command to prevent the console screen from flooding with messages and all calls from failing.

Sample output for some of these commands follows:

Router# debug ccsip events
CCSIP SPI: SIP Call Events tracing is enabled
21:03:21: sippmh_parse_proxy_auth: Challenge is 'Basic'.
21:03:21: sippmh_parse_proxy_auth: Base64 user-pass string is 'MTIzNDU2Nzg5MDEyMzQ1Njou'.
21:03:21: sip_process_proxy_auth: Decoded user-pass string is '1234567890123456:.'.
21:03:21: sip_process_proxy_auth: Username is '1234567890123456'.
21:03:21: sip_process_proxy_auth: Pass is '.'.
21:03:21: sipSPIAddBillingInfoToCcb: sipCallId for billing records =
10872472-173611CC-81E9C73D-F836C2B6@172.18.192.19421:03:21: ****Adding to UAS Request table

Router# debug ccsip info
00:20:32: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 172.18.207.10
:5060
00:20:32: CCSIP-SPI-CONTROL:  act_sentinvite_new_message
00:20:32: CCSIP-SPI-CONTROL:  sipSPICheckResponse
00:20:32: sip_stats_status_code
00:20:32: ccsip_get_code_class: !!Call Redirection feature is disabled on the GW
00:20:32: ccsip_map_call_redirect_responses: !!Mapping 302 response to 480
00:20:32:  Roundtrip delay 4 milliseconds for method INVITE