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

Models: H.323/SIP

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

Information About Reservationless Single Number Access Configuration

•[*]—Matches 0 or more occurrences of the preceding digit.

Step 5 Enter the type of prompt menu to play to the caller; then, enter a space. Valid selections are the following:

•CombinedAccess—Selects the Main menu.

•DIDMeeting—Prompts the caller for the meeting ID to join. This option can be used to place the caller directly into a meeting if the digits match an existing meeting ID on the Cisco Unified MeetingPlace Audio Server system.

•Profile—Prompts the caller for a profile number, which is not passed along to the Cisco Unified MeetingPlace server for user authentication.

•MeetingNotes—Prompts the caller to retrieve meeting notes.

Step 6 Enter the digits to present to the Cisco Unified MeetingPlace Audio Server system. Valid selections are the following:

•[0-9] [ A-D]—Presents the entered digits to the Cisco Unified MeetingPlace Audio Server system.

•KEEP—Preserves the dialed digits.

•NONE—Presents no digits to the server.

Step 7 Repeat Step 4 through Step 6 until the file contains one line for each dialing group that you want to configure.

Step 8 Save and close the dialgroups.txt file.

Step 9 Restart the IP-gateway server.

Configuring a Dialing Group Example

The following is a sample dialgroups.txt file that shows callers who dial extension 2121 are forwarded to meeting ID 656565. Callers who dial any other valid number are prompted to enter a profile number, and those digits are forwarded to the Cisco Unified MeetingPlace Audio Server system.

2121 DIDMeeting 656565

.* Profile KEEP

Information About Reservationless Single Number Access Configuration

With Reservationless Single Number Access (RSNA), profiled users who host or attend a reservationless meeting as either profile users or guests can access their meetings by dialing the same phone number, regardless of which Cisco Unified MeetingPlace Audio Server system is hosting the meeting. With RSNA, users always dial the number of their home server, which then transfers the call to the scheduler or host’s home server.

For information about configuring Reservationless Single Number Access, see the Administrator Guide for Cisco Unified MeetingPlace Audio Server Release 5.3 at the following URL:

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_maintenance_guides_list.html

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

 

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