Cisco Systems H.323/SIP manual MP Resp. Msg=3 CPerr=0 SeqNum=0x16

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Chapter 4 Troubleshooting Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Troubleshooting Caller Connectivity

Checking the Cisco Unified MeetingPlace Audio Server System When IP Ports Do Not Answer

Step 1 Check that the Ethernet switch port or any other network devices to which the MA-16 blade connects directly is set to fixed 100 Base-TX Full Duplex.

Step 2 Make sure that the IP ports on the server are configured and active by using the blade and portstat commands.

Step 3 Check the port status by performing the following steps:

a.Log in to the CLI.

b.At the tech$ prompt, enter the tvportstat -allcommand and monitor the output.

c.Make a test call.

d.Verify that the incoming call is seen by the server.

Step 4 Trace a test call by performing the following steps:

a.At the tech$ prompt, enter the cptrace -T 5 command and monitor the output.

b.Make another test call.

c.Verify that the incoming call is seen by the server.

Step 5 Check for warnings and alarms, especially those that occur in “cpiphandler.cc” by performing the following steps:

a.At the tech$ prompt, enter the viewexlog -s info -l more command.

b.Scroll through the log by entering f.

Step 6 At the tech$ prompt, enter gwstatus to verify that both the Cisco Unified MeetingPlace Gateway SIM and IP-gateway server have a status of OK.

Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server When IP Ports Do Not Answer

Step 1 To verify that both the Gateway SIM and IP-gateway server have a status of OK, enter gwstatus at the tech$ prompt.

Step 2 Verify that the Release 5.2.1 configuration has the appropriate call control enabled—either H.323 or SIP. Step 3 Open the Cisco Unified MeetingPlace Gateway SIM eventlog.

Step 4 Make a test call.

Step 5 From the Cisco Unified MeetingPlace Gateway SIM eventlog, verify that the test call is received by the IP-gateway server and that the call-processing server is returning a response code of 0, as shown the following example:

MP Resp. Msg=3 CPerr=0 SeqNum=0x16

Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

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OL-6571-02

 

 

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Contents Corporate Headquarters Text Part Number OL-6571-02Copyright 2005-2006 Cisco Systems, Inc. All rights reserved N T E N T S Fast Busy Signal Heard When Using a Cisco IP Phone Worksheets OL-6571-02 Audience ScopeNew Features in This Release Feature DescriptionNaming Conventions Used in This Guide Product Naming ConventionCisco Unified MeetingPlace System SIP RTP Protocol Description SIPAudio Quality During a Cisco Unified MeetingPlace Meeting Step Cisco IP Phone Description Pstn Phone Description Step Cisco IP Phone Description Pstn Phone Description Step Device Description Cisco SIP IP Phone Description Additional References Step H.323 Device Description Cisco SIP IP Phone DescriptionA P T E R CallManager Administration Field Description Click the Add a New Gateway linkChoose Originator Field Description Task Server to the Cisco Unified CallManager Configuration Tone boxChoose Start Run Configuring Cisco Unified MeetingPlace Gateway SIM Select MeetingPlace IP Gateway and click RemoveChanging Cisco Unified MeetingPlace Gateway SIM Settings To apply the configuration settings, click OK again OL-6571-02 A P T E R General Settings Setting Description DefaultSettings SIP Settings Field Name Setting SIP Field Name Setting Field Task Assigning the Primary IP AddressChoose Specify an IP address Information About Configuring a Dialing Group How to Configure a Dialing GroupConfiguring a Dialing Group Example OL-6571-02 Troubleshooting Network Connectivity Troubleshooting Caller Connectivity Unable to Make Calls From a Cisco IP PhoneDead Air Heard When Using a Cisco IP Phone Dead Air Heard When Using an H.323 DeviceFast Busy Signal Heard When Using a Cisco IP Phone MP Resp. Msg=3 CPerr=0 SeqNum=0x16 Page Unable to Dial Out on IP Ports MeetingPlace IP outdial. Phone=651515 IRC=0 PSTN=46 Unit=0 Troubleshooting Audio Problems Poor or Low-Audio QualityEcho OL-6571-02 Worksheets Description ValueDescription Value MeetingPlace IP call flow Value OL-6571-02 D E IN-2 IN-3 IN-4

H.323/SIP specifications

Cisco Systems has been a leading force in the development and implementation of voice and video communication technologies, prominently featuring H.323 and SIP (Session Initiation Protocol). These protocols have become cornerstones in the realm of IP-based communication, facilitating seamless interaction across disparate devices and networks.

H.323 is a set of protocols that provides multimedia communication over packet-switched networks, such as the Internet. It supports audio, video, and data communications across IP networks, utilizing a variety of components including terminals, gateways, and multipoint control units (MCUs). One key feature of H.323 is its ability to handle both point-to-point and multipoint video conferencing. This makes it particularly suitable for enterprise applications where group communication is essential.

On the other hand, SIP is a more recent and flexible signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video, and messaging applications. Renowned for its simplicity and interoperability, SIP can work with various communication mediums and provides extensive capabilities for managing multimedia sessions. One of the primary advantages of SIP is its scalability, allowing it to accommodate everything from small-scale personal communications to vast corporate systems.

Both H.323 and SIP support features such as call transfer, call hold, and caller ID. H.323, however, can be more complex due to its broader array of standards and components, which might require substantial configuration. SIP, conversely, is designed to be lightweight, easily integrated with existing systems and applications, making it more user-friendly for developers.

Cisco Systems enhances these protocols through their robust telecommunication infrastructure that facilitates performance optimization, security, and unparalleled user experience. With advanced technologies like Cisco Unified Communications Manager and Cisco Webex, organizations can leverage H.323 and SIP to create cohesive communication environments. Security features such as encryption and authentication ensure that sensitive conversations remain private and secure, while Quality of Service (QoS) protocols manage bandwidth effectively to maintain consistent call quality.

Ultimately, the combination of H.323's established framework and SIP's flexibility ensures that organizations using Cisco Systems can effectively manage their communication needs, fostering collaboration and connectivity in today's fast-paced digital landscape. These protocols continue to evolve, adapting to the ever-increasing demands placed on modern communication systems.
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