Cisco voip router models

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New and existing deployments can benefit by using any of these routers as unified communications gateways with Cisco Unified Communications Manager. Cisco, and and Series Integrated Services Routers can communicate directly with Cisco Unified Communications Manager, allowing for the deployment of unified communications solutions that are ideal for small and medium-sized businesses, large enterprises, and service Cisco voip router models that offer managed network services. These platforms provide a highly flexible and scalable solution for small and medium-sized branch and regional offices.

These platforms support a wide range of packet telephony-based voice interfaces and aling protocols within the industry, providing connectivity support for more than 90 percent of the world's private branch exchanges PBXs and public-switched telephone network PSTN connection points. You can configure these unified communications routers to support from 2 to voice channels.

Additionally, you can use these routers to terminate Session Initiation Protocol SIP trunking into the enterprise or branch office by enabling the Cisco Unified Border Element features. Additional details are available in the Cisco Unified Border Element data sheet. As your enterprise seeks to deploy an expanding list of unified communications applications and services, Cisco unified communications routers — interoperating with Cisco Unified Communications Manager — can provide a solution that will grow with your changing needs.

The gateway configuration and dial-plan configuration are centrally managed from Cisco Unified Communications Manager, which generates an XML config file that is downloaded by the gateway to autoconfigure. The Cisco, and Series unified communications routers can help you immediately deploy an end-to-end unified communications network architecture or gradually shift voice traffic from traditional circuit-switched networks to a single infrastructure carrying data, voice, and video over packet networks. Initially, you can use these unified communications routers to interconnect older PBXs over the packet infrastructure and still maintain PSTN off-net connectivity through your circuit-switched PBXs.

Later, you can migrate PSTN off-net connectivity to the unified communications routers and start to incorporate IP phones at larger sites Figure 1. After all sites are Cisco voip router models IP telephony, you can begin deploying IP-based applications such as IP unified messaging, personal assistants, and extension mobility.

Cisco voip router models

As companies seek to deploy unified communications solutions across the entire enterprise - converging voice, video, and data across potentially thousands of sites - they require a solution that offers simple administration, virtually unlimited scalability, and high availability. The unified communications routers work in concert with the Cisco Unified Communications Manager, deployed in either a distributed or centralized call-processing model, to provide the unified communications solutions that enterprises require.

Demand for technology to help increase employee productivity and reduce costs is at an all-time high. At the same time, many organizations are struggling to deploy new applications and services because of unavailable capital budgets. The centralized call-processing model can provide technology to users who require it, while simultaneously providing ease of centralized management and maintenance of applications to network administrators. This deployment model allows branch-office users to access the full enterprise suite of communications and productivity applications for the first time, while lowering total cost of ownership TCO.

There is no need to "touch" each branch office each time a software upgrade or new application is deployed, accelerating the speed in which organizations can adopt and deploy new technology solutions. The ability to quickly roll out new applications to remote users can Cisco voip router models a sustainable competitive advantage versus having to visit each of many branch-office sites to take advantage of new applications. An architecture in which a Cisco Unified Communications Manager and other Cisco IP Communications applications are located at the central Cisco voip router models offers the following benefits:.

As enterprises extend their IP telephony deployments from central sites to remote offices, an important consideration is the ability to cost-effectively provide failover capability at remote branch offices. However, the size and of these small-office sites preclude most enterprises from deploying dedicated call-processing servers, unified messaging servers, or multiple WAN links to each site to achieve the required high availability.

Cisco Unified Communications Manager with Survivable Remote Site Telephony SRST allows companies to extend high-availability IP telephony to their remote branch offices with a cost-effective solution that is easy to deploy, administer, and maintain. The router provides essential call-processing services for the duration of the failure, helping ensure that critical phone capabilities are operational.

Upon restoration of the connectivity to the Cisco Unified Communications Manager, the system automatically shifts call-processing functions back to the primary Cisco Unified Communications Manager cluster. Configuration for this capability is performed only once in the Cisco Unified Communications Manager at the central site Figure 2. Simple Administration. Table 1 summarizes the features of the unified communications routers with Cisco Unified Communications Manager. Table 2 summarizes roadmap unified communications features of these routers that are independent of Cisco Unified Communications Manager.

Please also refer to the Cisco Series Integrated Services Routers release notes for unified communications features added in different releases. Table 1. Cisco and Series ISRs. Analog FXS interfaces loop-start and ground-start aling. This aling facilitates direct connection to phones, fax machines, and key systems. These interfaces make direct connection to a PBX possible. Analog FXO interfaces loop-start and ground-start aling. Analog direct inward dialing DID. Road map. DSAPP 3-way conference. This feature enables an end user already engaged in a stable two-party call to add a third party to the conversation.

FXO tone answer supervision. This feature facilitates the use of tones to al answering a call and the start of a call detail record CDR. FXO disconnect supervision. This feature makes battery reversal or tones available for use to disconnect FXO calls. BRI Q. This feature enables connection to the PSTN. This feature enables connection to a PBX. SIG-basic call including calling Cisco voip router models. This feature facilitates connection to a PBX or key system. SIG forward, transfer, and conference. These services enable connection to a PBX or key system. T1-CAS feature group D 5.

These interfaces are used to connect Cisco voip router models a PBX or key system and to provide off-premises connections. These interfaces are used to connect to a PBX or key system. E1 CAS. E1 MelCAS. E1 R2 more than 30 country variants. SIG basic call including calling. This feature is used to connect to a PBX. SIG, including call diversion and forward, transfer, calling and connected ID services, and message-waiting indicator.

This feature helps assure high-ranking personnel communication to critical organizations and personnel during network stress situations. It allows priority calls for validated users to preempt lower-priority calls. Fractional PRI. Other channels are either unused or used for data. This feature is used to enable a connection trunk for common channel aling TCCS application.

SIG and Q. This feature integrates ISDN trunks with both voice and video traffic. Out-of-band dual-tone multifrequency DTMF. This feature carries DTMF tones and information out of band for clearer transmission and detection. High Availability.

Cisco voip router models

Cisco Unified Communications Manager failover redundancy. When the unified communications router loses contact with the primary Cisco Unified Communications Manager, the gateway uses the next available Cisco Unified Communications Manager. Cisco Unified Communications Manager call preservation during failover.

Existing calls are preserved during a failover to the next available Cisco Unified Communications Manager. Calls are also preserved upon restoration of the primary host Cisco Unified Communications Manager. SRST and gateway fallback. Gateway fallback provides support for PSTN telephony interfaces on the branch-office router for the duration of the loss.

Cisco voip router models

Existing calls are preserved during a loss of connection to the Cisco Unified Communications Manager cluster and gateway fallback. Calls are also preserved upon restoration of the Cisco Unified Communications Manager connection.

Music on Hold MoH. Multicast music on hold MoH - centralized. This feature helps the unified communications router deliver music streams from an MoH server to users on on- Cisco voip router models off-net Cisco voip router models. Multicast MoH - distributed. This feature helps the unified communications router deliver music streams to users through the router-embedded MoH server to on- and off-net calls. Tone on hold. Tone indicates when a user is placed on hold. Tone-on-hold timer tuning. Tone on hold is generated locally in the gateway for play to the PSTN.

Tone-on-hold timer tuning allows the use of service parameter settings in Cisco Unified Communications Manager for specification of the time between beeps. Caller ID. Caller ID support 8. This feature helps the unified communications router send the caller ID of a caller for display:.

In SIP and H. Group III fax support. This feature enables transmit T. Y- Cisco V. This feature delivers enhancements to the voice gateways to satisfy requirements outlined in the UCR specification. Specifically, support is added for the V. Modem relay. Modem relay demodulates a modem al at one voice gateway and passes it as packet data to another voice gateway where the al is remodulated and sent to a receiving modem. On detection of the modem answer tone, the gateways switch into modem passthrough mode and then, if the call menu CM al is detected, the two gateways switch into modem relay mode.

Modem passthrough. Modem passthrough over VoIP provides the transport of modem als through a packet network by using pulse code modulation PCM encoded packets. Off-ramp faxing allows a voice gateway that handles calls going out from the network to a fax machine or the PSTN to convert a fax with a TIFF attachment into a traditional fax format that can be delivered to a standard fax machine or the PSTN.

Standards-based codecs You can choose to transmit voice across your network as either uncompressed pulse code modulation PCM or compressed from 5. Voice activity detection VAD. VAD conserves bandwidth during a call when there is no active voice traffic to send. Comfort-noise generation. While using VAD, the digital al processor DSP at the destination end Cisco voip router models background noise from the source side, preventing the perception that a call is disconnected.

Private-line automatic ringdown PLAR. PLAR provides a dedicated connection to another extension or an attendant.

Cisco voip router models

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Introducing VoIP Gateways (Introducing Voice over IP Networks) Part 1