The Cisco Unified Phone Designer is a downloadable Cisco Unified Communications widget that I will discuss in this blog. Read more
Cisco’s second generation integrated services router (ISR G2) introduces new digital signal processors (DSP) with up to four times the capacity of the Packet Voice DSP Modules used in the 2800 and 3800 series ISR routers. The 2800 and 3800 series router use PVDM2 (2nd generation) DSP modules on the motherboard of the router, while the 2900 and 3900 series ISR G2 routers ship with PVDM3 slots on the motherboard. Read more
In this blog, we will take a look at the Services Performance Engine (SPE) supported on the Cisco 3900 series router. The SPE is a modular motherboard design that allows customers to upgrade their routers core functionality without performing a forklift upgrade. The concept is very similar to the modularity that has allowed the following routers and switches to persist for close to or over a decade:
•7200 series router input/output (I/O) module Read more
•7500 series router – route switch processor (RSP) / versatile interface processor (VIP)
This blog will discuss the Power over Ethernet (PoE) service modules (SM) supported in the second generation Cisco Integrated Services router (ISR G2). Read more
The Cisco IOS gateway class of service configuration is accomplished using corlist configurations in the gateway. This blog is a continuation of the last CORLIST blog.
The incoming corlist (keyring) configuration to meet the class of service requirements is as follows:
Dial-peer cor list lobby_cos Read more
Member e911
!
Dial-peer cor list intern_cos
Member e911
Member local
!
Dial-peer cor list employee_cos
Member e911
Member local
Member longdistance
!
Dial-peer cor list manager_cos
Member e911
Member local
Member longdistance
Member international
In this blog, I will explain the Cisco IOS technique of applying class of service (CoS) restrictions. Previous blogs have covered calling search space (CSS) and partition configuration in Cisco Unified Communications Manager (CUCM) which will handle CoS call restrictions during normal operation of the network. Branch sites using Cisco Unified Communications Manager Express (CUCME) must rely on the Class or restriction list (CORLIST) functionality that is discussed in this blog. Read more
Cisco announced the ISR G2 routers today (integrated services router - second generation). This blog interrupts the SRST blog series, but I will get back to that conversation very soon. Let's take a look at the ISR G2 routers...
The ISR G2 router families include the following models listed from smallest to largest form factor:
• 800 Series Routers (860, 880, 890) Read more
• 1940 Series Routers (1941, 1941W)
• 2900 Series Routers (2901, 2911, 2921, 2951)
• 3900 Series Routers (3925, 3945)
Cisco announced the new Cisco ISR G2 routers today (integrated services router - second generation). This blog interrupts the SRST blog series, but I will get back to that conversation very soon. We will take a look at the ISR G2 routers in this blog.
The ISR G2 router families include the following models listed from smallest to largest form factor:
• 800 Series Routers (860, 880, 890) Read more
• 1940 Series Routers (1941, 1941W)
• 2900 Series Routers (2901, 2911, 2921, 2951)
• 3900 Series Routers (3925, 3945)
In this blog, we will investigate redirected dialed number identification service (RDNIS) to aid our understanding of the requirement when forwarding calls from a branch site location to a centralized voicemail solution when the WAN link is down.
Let’s take a look at an RDNIS example… We will discuss the calling (ANI) and called (DNIS) party information transmitted for a call forwarded to voicemail. Read more
In this blog, we will discuss SRST voicemail capabilities. The information in this blog assumes some information from previous blog postings on SRST. This SRST blog is the eighth SRST post in the SRST blog series. Read more
This blog will continue the SRST conversation that we have been having over the last six blogs. During normal operation, most implementations of Cisco Unified Communications Manager (CUCM) provide a secondary (stutter) dial tone after the access code of 9 is dialed. The 911 route pattern must be configured to provide an outside dial tone in addition to all the other route patterns beginning with a 9. Read more
In this blog, we will continue discussing more configuration options available in Survivable Remote Site Telephony (SRST). All of the configuration command examples in this blog are configured in call-manager-fallback configuration mode. Read more
In this blog, we will discuss SRST in more detail. Remote site MGCP gateways must be provisioned with MGCP fallback to route calls to and from the PSTN when the MGCP call agent (CUCM) is unreachable. This blog will cover the options to configure TDM call routing over a remote site gateway in SRST mode. Read more
MGCP fallback will allow inbound call routing over any MGCP controlled endpoint (TDM voice port).
The following configuration example is a simple Survivable Remote Site Telephony (SRST) configuration:
Router(config)#call-manager-fallback
Router(config-cm-fallback)#ip source address 10.1.1.1 port 2000
Router(config-cm-fallback)#max-ephones 30
Router(config-cm-fallback)#max-dn 70 octo-line
The call-manager-fallback global configuration command accesses SRST configuration mode. This command is very similar to the telephony-service configuration command used to access Cisco Unified Communications Manager Express (CUCME) configuration mode. Read more
MGCP gateways that lose connectivity to their primary Cisco Unified Communications Manager (CUCM) support call survivability by default, but the MGCP gateway cannot be used for any new inbound or outbound calls during a CUCM outage. During the CUCM outage, phones with existing calls will see an LCD message indicating CUCM is down and features are disabled (supplementary services). The MGCP gateway will re-register to a backup CUCM server when all calls have gone through normal call clearing at the end of a phone call. Read more
This blog will discuss the Cisco SRST conversation that was started in the last blog. After a Cisco IP phones registers to the Cisco SRST gateway, the Cisco IP phones will continue to send keep alive messages to the primary Cisco Unified Communications Manager (CUCM) server. When the primary CUCM server is available, the connection monitor duration timer must expire before the phone re-registers back to the primary CUCM server. The connection monitor duration default is 120 seconds by default, but can be configured in the CUCM device pool configuration. Read more
Survivable Remote Site Telephony (SRST) is a Cisco Unified Communications Manager (CUCM) call processing backup mechanism that allows Cisco IP phones to register to a Cisco router. A Cisco router in SRST mode provides call setup and call teardown services similar to CUCM, but only when connectivity to the CUCM servers provided in the Cisco Call Manager group are not reachable. Read more
Network time protocol (NTP) is a mechanism used to synchronize the date and time over IP based networks. Cisco Unified Communications platforms have a built in NTP client to ensure the proper date and time on the Cisco IP phones LCD display, call detail records (CDR), call management records (CMR), and trace files used for troubleshooting. Date and time information is automatically replicated to all subscribers in the cluster. Some UC platforms now support a similar database replication model as the publisher/subscriber model used in Cisco Unified Communication Manager (CUCM). Read more
Cisco Unified Communications Manager (CUCM) 5.0 introduced presence support. Presence is similar to the busy station select, busy lamp field functionality used in some traditional PBX system. Standards based presence information can be shared amongst different vendors using SIP trunks in CUCM. It is very common to share presence information with a Microsoft OCS server in environments running the Microsoft Office Communicator (MOC) client. This blog will provide an overview of presence and describe the functionality of the subscribe calling search space (CSS) configuration element. Read more
An AAR calling search space (CSS) is a special application of calling search spaces that can be applied at the line (directory number) level of Cisco IP phone configuration. This blog will discuss automated alternate routing (AAR) and the application of the AAR CSS.
AAR technology allows Cisco IP phone to Cisco IP phone calls over the WAN to be re-routed to the public switched telephone network (PSTN) when the configured location-based bandwidth of the location is in use.
AAR assumes the following configurations and technologies:
• Centralized Call Processing Read more
Dennis Hartmann, CCIE No. 15651, is a consultant with www.highpoint.com and author of Implementing Cisco Unified Communications Manager, Part 1. Dennis is also a lead instructor at Global Knowledge. Dennis has various certifications, including the Cisco CCVP, CCSI, CCNP, CCIP, and the Microsoft MCSE. Dennis has various specializations including unified communications, data center, routing & switching, service provider (MPLS and optical). Dennis has worked for various Fortune 500 companies, including AT&T, Sprint, Merrill Lynch, KPMG, and Cabletron Systems. He lives with his wife and children in Hopewell Junction, New York.
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