Cisco Systems H.323/SIP manual Troubleshooting Caller Connectivity

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

Troubleshooting Caller Connectivity

Troubleshooting Caller Connectivity

•Unable to Make Calls From a Cisco IP Phone, page 4-2

•Unable to Call a PSTN Telephone From a Cisco IP Phone or Vice Versa, page 4-2

•Dead Air Heard When Using an H.323 Device, page 4-3

•Dead Air Heard When Using a Cisco IP Phone, page 4-3

•Fast Busy Signal Heard When Using a Cisco IP Phone, page 4-3

•Unable to Make Dial-Pad Key Selections When Using an H.323 Device, page 4-3

•Checking the Cisco Unified MeetingPlace Audio Server System When IP Ports Do Not Answer, page 4-4

•Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server When IP Ports Do Not Answer, page 4-4

•Checking Cisco Unified CallManager When IP Ports Do Not Answer, page 4-5

•Checking the Cisco Unified MeetingPlace Audio Server System When IP Calls Connect But No Audio Is Heard, page 4-5

•Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 When IP Calls Connect But No Audio Is Heard, page 4-6

•Checking the Cisco IP Phone When IP Calls Connect But No Audio Is Heard, page 4-6

•Unable to Dial Out on IP Ports, page 4-6

•Checking the Cisco Unified MeetingPlace Audio Server System When Unable to Dial Out on IP Ports, page 4-7

•Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server When Unable to Dial Out on IP Ports, page 4-7

•Checking Cisco Unified CallManager When Unable to Dial Out on IP Ports, page 4-8

Unable to Make Calls From a Cisco IP Phone

Possible Cause—The network may not be functioning properly.

Corrective Action—Verify your network access.

Possible Cause—Cisco Unified CallManager may not be configured correctly.

Corrective Action—Verify your Cisco Unified CallManager configuration.

Unable to Call a PSTN Telephone From a Cisco IP Phone or Vice Versa

Possible Cause—The voice gateway may not be functioning or configured properly.

Corrective Action—Verify your configuration settings.

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 ScopeNaming Conventions Used in This Guide New Features in This ReleaseFeature Description 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 Choose Originator Click the Add a New Gateway linkField Description 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 Settings Setting Description DefaultGeneral Settings SIP Settings Field Name Setting SIP Field Name Setting Choose Specify an IP address Assigning the Primary IP AddressField Task 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 PhoneFast Busy Signal Heard When Using a Cisco IP Phone Dead Air Heard When Using an H.323 DeviceDead Air 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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