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ネットワークセキュリティやってます。技術よりも趣味と雑談が多めのブログです。最近はオンライン英会話にはまっています。

2017-09

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[07/24 hechtia]
[06/23 Stream.T]
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■ ITU-T Recommendation H.323 describes an infrastructure of terminals, common control components, services, and protocols that are used for multimedia communications.
■ Functional components of H.323 include terminals, gateways, gatekeepers, Cisco UBEs, and MCUs.
■ Calls can be established between endpoints, endpoints to gatekeepers, or gatekeepers to gatekeepers.
■ H.323 calls can occur with or without the use of a gatekeeper.
■ H.323 defines three types of multipoint conferences.
■ When configuring codecs, you can specify one codec or set up codec negotiation.
■ You might want to adjust some of the H.323 timers to meet network requirements.
■ You can use several commands to configure fax features on H.323 gateways.
■ DTMF relay solves the problemof DTMF distortion.
■ Use the show gateway command to verify H.323 gteway status.
■ MGCP defines an environment for controlling telephony gateways from a centrralized call agent.
■ MGCP components include endpoints, gateways, and call agents.
■ Calls and connections are basic concepts in MGCP.
■ MGCP call flow consists of an exchange of messages between a call agent and a gateway.
■ The mgcp command can be used to configure residential and trunk gateways on a Cisco router.
■ Several show and debug commands help to verify an MGCP configuration.
■ SIP is defined by IETF RFCs 2543 and 3261 and allows integration with third-party VoIP networks.
■ SIP is modeled on the interworking of UAs and network servers.
■ A SIP call flow consists of signaling and transmission of bearer and media packets.
■ Communication between SIP components uses a request and response message model.
■ A SIP address consists of an optional user ID, a host description, and optional parameters to qualify the address more precisely.
■ SIP call setup models include direct, proxy server, and redirection.
■ You can use several commands on Cisco IOS to configure SIP on Cisco IOS routers.
■ You can use several commands on Cisco IOS to verify and troubleshoot a SIP integration.

PR

Cisco Unified CMのインストールの仕方とか、つまらなすぎる。

CUCMの英語マニュアルの読み上げをひたすら聴くという睡魔との闘いになっている。


a8614458.jpgd1849856.jpg







 

次は面白くないCIPT1を我慢して聴いてみる。

88b7f548.jpg









the CVOICE On-Demand marker
10/7 1A
10/8 1B
10/9 1end
10/10 2B
10/11 2C
10/12 2D-33
10/14 2end
10/15 3A
10/16 3end
10/18 4A
10/19 4B
10/20 4C
10/22 4D
10/24 5intro
10/25 5A
10/26 5end
10/27 6A
10/28 Over
■ Digital voice ports are found at the intersection of a packet voice network and a digital, circuit-switched telephone network.

■ T1 CAS use a digital T1 circuit together with in-band CAS.

■ E1 digital circuits can be deployed using R2 signaling.

■ ISDN is a circuit-switched telephone network system designed to allow digital transmission of voice and data over ordinary telephone copper wires.

■ ISDN uses Q.921 and Q.931 for signaling.

■ Before configuring a T1 or E1 trunk, you must gather information about the requirements for Framing, Linecode and DS0 gourps.

■ Configuring an E1 trunk is similar to configuring a T1.

■ Many PBX vendors support either T1/E1 PRI or BRI connections.

■ Various show commands are available to verify and monitor digital voice ports.

■ QSIG allow feature transparency between different vendor PBXs.

■ QSIG can be configured over PRI or BRI.

■ Various show and debug commands are available to verify the QSIG connection.

■ A local call is handled entirely by the router and does not travel over an external network.

■ On-net calls can be routed through one or more voice-enabled routers, but the calls remain on the same network.

■ An off-net call occurs when a user dials an acces code from a telephone directly connected to a voice-enabled router or PBX to gain access to the PSTN.

■ Voice port call types include local, on-net, off-net, PLAR, PBX to PBX, intercluster trunk, and on-net to off-net calls.

■ Voice ports on routers and access servers emulate physical telephony switch connections.

■ Analog voice port interfaces connect routers in packet-based networks to analog tow-wire or four-wire analog circuits in telephony networks.

■ FXS, FXO, and E&M ports have several configuration parameters.

■ CAMA is used for 911 and E911 services.

■ DID service enalbes callers to dial an extension directly on a PBX or packet voice system.

■ You can set a number of timers and timing parameters for fine-tuning a voice port.

■ The show, debug, and test commands are used for monitoring and troubleshooting voice functions in the networks.

■ Dial peers are used to identify call source and destination endpoints and to define the characteristics applied to each call leg in the call connection.

■ An end-to-end voice call consists of four call legs.

■ A dial peer is an addressable call endpoint.

■ POTS dial peers retain the characteristics of a traditional telephony network connection.

■ When a matching inbound dial peer is not found, the router resorts to the default dial peer.

■ The destination pattern associates a telephone number with a given dial peer.

■ When determining how inbound dial peers are matched on a oruter, it is important to note whether the inbound call leg is matched to a POTS or VoIP dial peer.

■ Outbound dial-peer matching is completed on a digit-by-digit basis.
著書




プロフィール
HN:
ぜん吉
性別:
男性
職業:
割と自由なリーマン
趣味:
海外出張
自己紹介:
2006年のCCNA合格を皮切りにCCIE-RSを含めて数々のシスコ資格をパスし、2009年に念願の海外受験(ドバイ)でCCIE-SCを取得。そして、2010年に目標だったトリプルCCIEを香港の地にて達成した。今はネットワークセキュリティやデータ分析などをやっています。

■2006年の目標
CCNA(達成)

■2007年の目標
CCNP(達成)
CCDA(達成)
CCDP(達成)
CCIP(達成)

■2008年の目標
CCSP(達成)
CCIE-RS(達成)
TOEIC700点(達成)

■2009年の目標
CCIE-Sec(達成)
TOEIC800点(達成)
JNCIA-JUNOS(達成)

■2010年の目標
JNCIA-M(達成)
CCIE-SP(達成)
JNCIS-M(達成)
JNCIA-EX(達成)
JNCIS-SEC(達成)

■2011年の目標
異動(未達成)

■2012年の目標
異動(未達成)
TOEIC850点(達成)

■2013年の目標
異動(達成)
CCIE更新(達成)

■2015年の目標
本を出す(達成)

■2017年の目標
TOEIC900(達成)

■2018年の目標
海外勤務








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