SIP Troubleshooting

This topic describes how to troubleshoot the SIP integration of Cisco Unity Express and Cisco Unified Communications Manager Express.

  • Cisco Unified Communications Manager Express and Cisco Unity Express communicate via SIP.
  • Voicemail integration issues require troubleshooting SIP between these systems.
  • Verify that SIP configuration parameters, such as IP address, protocol version (SIP version 2), and dual tone multifrequency settings, are the same for both systems.

Cisco Unified Communications Manager Express and Cisco Unity Express communicate via SIP. This method of communication means that voicemail integration issues require troubleshooting SIP between these systems. Verify that the SIP configuration parameters, such as IP address and protocol version (SIP version 2), are the same for both systems.

SIP Call Flow

The figure shows a SIP call flow between Cisco Unified Communications Manager Express and Cisco Unity Express.

  • SIP INVITE message initiates the call setup between Cisco Unified Communications Manager Express and Cisco Unity Express.
  • The message direction depends on which side originates the call.
  • SIP 200 OK is sent when the call is answered.

SIP uses different message types to initiate and control voice calls. The SIP request messages are as follows:

  • INVITE: This message indicates that a user or service is being invited to participate in a call session.
  • ACK: This message confirms that a client has received a final response to an INVITE request.
  • BYE: This message terminates an existing call, and either user agent can send the message.
  • CANCEL: This message cancels pending searches, but it does not terminate calls that have been accepted.
  • OPTIONS: This message queries the capabilities of servers.
  • REGISTER: This message registers the user agent with the registrar server of a domain.

The SIP response messages are numbered and grouped in the following ranges:

  • 1xx: Information
  • 2xx: Successful responses
  • 3xx: Redirection responses
  • 4xx: Request failure responses
  • 5xx: Server failure responses
  • 6xx: Global responses

The SIP INVITE message initiates the call setup between Cisco Unified Communications Manager Express and Cisco Unity Express. The INVITE message direction depends on which side originates the call. The SIP 200 OK message is sent when the call is answered.

The following summarizes a successful SIP call setup:

  • Send INVITE message.
  • Receive 100 Trying.
  • Receive 180 Ringing.
  • Send 200 OK.
  • Receive ACK.

The following summarizes a successful SIP call clearing:

  • Send BYE message.
  • Receive 200 OK.
  • Receive ACK.
Note

Refer to RFC 3261 for more information about the SIP message types.

Common SIP configuration problems are the following:

  • Incorrect IP address for Cisco Unity Express
  • Incorrect codec configuration
  • Wrong destination pattern is configured, so the dial peer is not triggered for voicemail calls
  • Misconfigured DTMF relay
  • Incorrect SIP version
  • VAD is enabled by default; it must be disabled

To enable Cisco Unity Express SIP trace options by using the GUI, navigate to the Cisco Unity Express GUI, choose Administration > Traces, and check the caff-sip macro check box.

SIP messages are the same as messages that are viewed by using the Cisco IOS Software debug command. Additional internal Cisco Unity Express messages may also be displayed in trace output.

Note

Trace output is viewed via the CLI. For example, use the show trace buffer tail command to view trace output in real time.

SIP Issue Troubleshooting

This topic describes a SIP troubleshooting issue.

  • A company is using a Cisco Unity Express voicemail solution.
  • Currently, all users are experiencing problems accessing their voicemail.
  • Users report the following when pressing the Messages button:
    1. They hear a busy tone.
    2. The Unknown Number message is displayed on the phone.

In this SIP troubleshooting scenario, a company uses a Cisco Unity Express voicemail solution with Cisco Unified Communications Manager Express. All users are experiencing problems accessing their voicemail. What may be the reason?

  • The administrator suspects a problem with the voicemail system or an integration issue.
    1. All voicemail users experience the issue.
  • No issues are found during initial verification of the configurations.
  • The next step is to look at the connection between Cisco Unity Express and Cisco Unified Communications Manager Express.
    1. Use Cisco Unity Express SIP tracing to view traffic.

All voicemail users experience this issue, so the administrator suspects a problem with the voicemail system or an integration issue. The first action is to verify the Cisco Unified Communications Manager Express and Cisco Unity Express configurations. In this example, no issues are found during the initial verification of the configurations and integration between Cisco Unified Communications Manager Express and Cisco Unity Express.

The next step is to look at the connection between Cisco Unity Express and Cisco Unified Communications Manager Express. Use Cisco Unity Express SIP tracing to view traffic between Cisco Unified Communications Manager Express and Cisco Unity Express. Alternatively, use the Cisco Unified Communications Manager Express debug command to view the SIP traffic.

In this example, the codec was configured incorrectly. When debugging or tracing, take a look at the SIP INVITE message. The INVITE will contain SDP parameters for the call setup. In the trace, the INVITE and 200 OK SIP messages will have different codec types as follows:

  • INVITE requests G.729: a = rtpmap:18 G729/8000 0
  • 200 OK requests G.711 mu-law: a = rtpmap: 0 PCMU/8000

In a trace, the BYE message will contain a reason code for the call disconnection (cause = 65). Cause 65 indicates “bearer capability not implemented,” which indicates the wrong codec.

Additional and commonly seen cause codes include the following:

  • Unallocated (unassigned) number = 1
  • Normal call clearing = 16
Note

See the ITU Q.850 standard for a complete list of cause codes.

In this example, the dial peer is not configured to use the G.711 mu-law codec, so the default G.729 will be used. The voicemail dial peer must be configured for G.711 mu-law using the codec dial-peer configuration command to resolve the issue.

Advertisements

Author: drbabbers

ccieme.wordpress.com - my personal journey to ccie