Cisco Systems H.323/SIP manual Information About Configuring a Dialing Group

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Chapter 3 Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Information About Configuring Multiple Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Step 5 Click OK to apply your settings and return to the desktop.

Information About Configuring Multiple Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Servers for Load Balancing and Redundancy

If you have deployed multiple IP-gateway servers to route IP calls, you can configure Cisco Unified CallManager or your IP PBX to load balance and to provide Cisco Unified MeetingPlace system redundancy by creating route groups that send calls to other IP-gateway servers if gateway failure occurs. A route group allows you to designate the order in which IP-gateway servers are selected and to prioritize a list of IP-gateways and ports for outgoing trunk selection.

All IP-gateway servers actively handle calls, and calls are routed round-robin among the IP-gateway servers. Therefore, in-session calls that are connected to a IP-gateway server that has failed are disconnected, and those callers must call again to be reconnected to the Cisco Unified MeetingPlace Audio Server system. New callers, however, are routed to another IP-gateway server.

For information about configuring route groups, see to the Redundancy Chapter in the Cisco Unified CallManager System Guide for your software release at the following URL:

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/index.htm

Information About Configuring a Dialing Group

Dialing groups customize the Cisco Unified MeetingPlace Audio Server system by presenting specific voice prompts to callers who dial in to a meeting by using a particular IP phone number. For example, you can configure a dialing group to immediately place callers who dial extension 2121 into meeting ID 656565.

You configure dialing groups by editing the dialgroups.txt file to include the dial pattern with which to associate a specific dialing group; the application, or prompt, to play for the dialing group callers; and the meeting number to present to the Cisco Unified MeetingPlace Audio Server system. Entries in dialgroups.txt are processed in order from top to bottom. If a match is not found, the caller is placed at the CombinedAccess menu, and the dialed digits are presented to the Cisco Unified MeetingPlace Audio Server system.

How to Configure a Dialing Group

Step 1 Open the Cisco Unified MeetingPlace IP Gateway folder on your IP-gateway server.

Step 2 By using a text editor, open the dialgroups.txt file.

Step 3 Read the comment lines that start with the # symbol.

Step 4 Enter the dial pattern that you want to customize; then, enter a space. Valid selections are the following:

[0-9] [ A-D]—Presents the digits to the MeetingPlace audio server.

[.]—Matches any valid digit.

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 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.