Image 1 - Inbound and Outbound Call Legs IllustratedĪ successful call through a Cisco Gateway ALWAYS* matches an inbound or outbound dial-peer to route properly. Image 1 shows a call from the PSTN to CUCM routing through a Cisco Voice Gateway and the respective inbound and outbound call leg information. The gateway processes the call and based on its processing sends the call to the next call agent. Scenario: A call arriving at a Cisco gateway from another call agent is the inbound call leg (in-leg).
There are far too many Call Agents to list. This can be and is not limited to Telephony Provider equipment, a Cisco Gateway, an IP Phone, a Cisco Unified Communication Manager (CUCM), Cisco Unity connection (CUC), etc. A call agent is a device that initiates, processes, or forwards telephony calls. A call leg is the bidirectional communication between two call agents. This is the device that stands as the demarcation point between a customer LAN and an ITSP/PSTN NetworkĬisco IOS and IOS-XE gateways utilize a concept of a dial-peer to control call routing and capabilities negotiation for each leg of a call. URI CSCua14749 Carrier-ID does not work on IOS-XE platformsĪ Cisco Proprietary Header for ILS Route-Strings used with SIP.ĮNUM is a protocol that uses Domain Name Service (DNS) to translate E164 phone numbers into URIs. This is the Calling Number or the Originating Calling Number for a call.This can also be referred to as the Calling Line Identifier (CLID) which can also be titled the Caller IDĪ URI is either a sip: or tel: string most commonly used with VoIP Protocols SIP and H323. This is the Called Number or the Destination Number for a call.
Uri sip definition plus#
Common Definitions AttributeĪlso referred to as number string, phone number, number, or E164 number.Consists entirerly of digits 0 through 9 with an optional leading plus symbol (+).ĭialed Number Identification Service (DNIS) If your network is live, ensure that you understand the potential impact of any command.
All of the devices used in this document started with a cleared (default) configuration. The information in this document was created from the devices in a specific lab environment.
Uri sip definition software#
The information in this document is based on these software and hardware versions: Media Protocols: Real Time Protocol (RTP), voice codecs, video codecs.Īnalog Technologies: Ear and Mouth (E&M), Foreign Exchange Subscriber (FXS) and Foreign Exchange Office (FXO). Signaling Protocols: Session Initiation Protocol (SIP), H323 (h225 / h245), Media Gateway Control Protocol (MGCP), Skinny Client Control Protocol (SCCP), ISDN Q931, E1 R2. These protocols are referenced many times throughout the book. While there are no formal prerequisites needed to read this document, it is written with the expectation that the reader already has some working knowledge of underlying voice signaling protocols that are used to establish and connect phone calls.
If there is a very notable defect it is linked within the text so that readers are aware.Ĭontributed by Kyzer Davis and edited by Ramiro Amaya, Cisco TAC Engineers Prerequisites Requirements This information can also be referenced quickly by checking the Command and Feature Roadmap section. The many features in this document are clearly marked with the version the feature was introduced to both IOS and IOS-XE. This document utilizes configuration examples as well as debug and show command outputs as reference points. These include digit manipulation, a quick overview of Session Initiation Protocol (SIP) message manipulation, a few methods for restricting calling capabilities, a quick media and signaling binding overview, and lastly a bit of troubleshooting. In addition to dial-peer information this document covers important topics that pertain to call routing. The purpose of this document is to explain IOS and IOS-XE Call Routing and the mechanisms behind inbound and outbound dial-peer matching with Plain Old Telephone Service (POTS) and Voice over IP (VoIP) Network call legs.
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