If calls cannot be made between SIP gateways or over SIP trunks, dial peer configuration is one of the first places to check. Make sure that the dial peer is configured to use SIP and that both devices are using the same transport protocol and DTMF relay method. Make sure that destination patterns and session targets are correct, also.
Several show commands can troubleshoot and monitor the SIP UA function of the gateway. Example 4-12 lists them; options can vary by Cisco IOS and device.
[View full width] SIP-GW#show sip-ua ? calls Display Active SIP Calls connections Display SIP Connections map Display SIP status code to PSTN cause mapping table & vice versa min-se Display Min-SE value mwi Display SIP MWI server info register Display SIP Register status retry Display SIP Protocol Retry Counts service Display SIP submode Shutdown status statistics Display SIP UA Statistics status Display SIP UA Listener Status timers Display SIP Protocol Timers |
The show sip-ua connections {udp|tcp} command gives you information on active connections, including those with errors. To stop a problem connection, use the clear sip-ua {udp | tcp} [connection id number] [target ipv4:ip-address] command.
To ensure that the SIP is enabled on the gateway, use the show sip-ua service command. You should get the following result:
SIP-GW#show sip-ua service SIP Service is up
The show sip-ua statistics command provides statistics on each type of method and response, errors, and total SIP traffic information. You can reset these counters with the clear sip-ua statistics command.
The show sip-ua status command can be useful in troubleshooting, also. Output from this command was shown previously in Example 4-13.
To debug SIP messages, use the debug ccsip command. This command has several options, as Example 4-13 shows. Use messages to see the SIP method and response messages, as shown previously in Example 4-1. The media option shows RTP information. Your options might vary by Cisco IOS and device.
SIP-GW#debug ccsip ? all Enable all SIP debugging traces calls Enable CCSIP SPI calls debugging trace error Enable SIP error debugging trace events Enable SIP events debugging trace info Enable SIP info debugging trace media Enable SIP media debugging trace messages Enable CCSIP SPI messages debugging trace preauth Enable SIP preauth debugging traces states Enable CCSIP SPI states debugging trace transport Enable SIP transport debugging traces |
Part I: Voice Gateways and Gatekeepers
Gateways and Gatekeepers
Part II: Gateways
Media Gateway Control Protocol
H.323
Session Initiation Protocol
Circuit Options
Connecting to the PSTN
Connecting to PBXs
Connecting to an IP WAN
Dial Plans
Digit Manipulation
Influencing Path Selection
Configuring Class of Restrictions
SRST and MGCP Gateway Fallback
DSP Resources
Using Tcl Scripts and VoiceXML
Part III: Gatekeepers
Deploying Gatekeepers
Gatekeeper Configuration
Part IV: IP-to-IP Gateways
Cisco Multiservice IP-to-IP Gateway
Appendix A. Answers to Chapter-Ending Review Questions
Index