Example Nyquist Theorem

While the human ear can sense sounds from 20 to 20,000 Hz, and speech encompasses sounds from about 200 to 9000 Hz, the telephone channel was designed to operate at about 300 to 3400 Hz. This economical range carries enough fidelity to allow callers to identify the party at the far end and sense their mood. Nyquist decided to extend the digitization to 4000 Hz, to capture higher-frequency sounds that the telephone channel may deliver. Therefore, the highest frequency for voice is 4000 Hz, or...

Data Link Overhead

This topic lists overhead sizes for various Layer 2 protocols. This topic lists overhead sizes for various Layer 2 protocols. Another contributing factor to bandwidth is the Layer 2 protocol used to transport VoIP. VoIP alone carries a 40-byte IP UDP RTP header, assuming uncompressed RTP. Depending on the Layer 2 protocol used, the overhead could grow substantially. The larger the Layer 2 overhead, the more bandwidth required to transport VoIP. The following points illustrate the Layer 2...

OnNet to Off Net Call

All rights re When planning a resilient call-routing strategy, it may be necessary to reroute calls through a secondary path should the primary path fail. On-net to off-net calls originate on an internal network and are routed to an external network, usually to the PSTN. On-net to off-net call-switching functionality may be necessary when a network link is down, or if a network becomes overloaded and unable to handle all calls presented.

What Is Jitter

Jitter is defined as a variation in the delay of received packets. The sending side transmits packets in a continuous stream and spaces them evenly apart. Because of network congestion, improper queuing, or configuration errors, the delay between packets can vary instead of remaining constant, as shown in the figure. This variation causes problems for audio playback at the receiving end. Playback may experience gaps while waiting for the arrival of variable delayed packets. When a router...

How Does Cisco Call Manager Express Cisco Unity Express Work Cont

All rights re PSTN Gateway and IP to IP Gateway PSTN Gateway and IP to IP Gateway 2005 Cisco Systems, Inc. All rights re If the Cisco CME system needs to set a call up to an IP phone under the control of another CME system, then the H.323 protocol will need to be used between the Cisco CME systems. This allows for many different deployments of Cisco CME to be integrated together through an IP-based WAN link. The PSTN gateway function can be performed on the Cisco CME...

VoIP Dial Peer Configuration

Configuring Destination-Pattern Options This topic describes destination-pattern options and the applicable shortcuts. This topic describes destination-pattern options and the applicable shortcuts. The destination pattern associates a telephone number with a given dial peer. The destination pattern also determines the dialed digits that the router collects and forwards to the remote telephony interface, such as a PBX, Cisco CallManager, or the PSTN. You must configure a destination pattern for...

System Display Message Idle URL

The display of the IP phone can be customized to reflect the needs and identity of the enterprise that the CallManager system and phones are being deployed for. Normally, the IP phone header bar, or top line, of a 7940 or 7960 phone replicates the text that appears next to the first line button. The header bar can, however, contain a user-definable message instead of the extension number. For example, the header bar can be used to display a name or the full E.164 number of the phone. If no...

Example Matching Outbound Dial Peers

In the figure, dial peer 1 matches any digit string that has not matched other dial peers more specifically. Dial peer 2 matches any seven-digit number in the 30 and 40 range of numbers starting with 55501. Dial peer 3 matches any seven-digit number in the 20 range of numbers starting with 55501. Dial peer 4 matches the specific number 5550124 only. When the number 5550124 is dialed, dial peers 1, 3, and 4 all match that number, but dial peer 4 places that call because it has the most specific...

Files Cont Default XML File

< callManagerGroup> < members> < member priority M0> < callManager> < ports> < loadInformation6 model IP Phone < loadInformation124 model Addon < loadInformation9 model IP Phone 7935> < loadInformation9> < loadInformation8 model IP Phone < loadInformation7 model IP Phone < loadInformation20000 model IP Phone < loadInformation30008 model IP Phone < loadInformation30002 model IP Phone < loadInformation30019 model IP Phone < loadInformation30007 model...

Types of Trunk Signaling

This topic describes the trunk and line-seizure signaling types. This topic describes the trunk and line-seizure signaling types. There must be signaling standards between the lines and trunks of a telephone network, just as there are signaling standards between a telephone and the telephone company. Trunk signaling serves to initiate the connection between the switch and the network. There are five different types of trunk signaling and each applies to different kinds of interfaces, such as...

Foreign Exchange Station Interface

This figure depicts an FXS interface. The FXS interface provides a direct connection to an analog telephone, a fax machine, or a similar device. From a telephone perspective, the FXS interface functions like a switch therefore, it must supply line power, ring voltage, and dial tone. The FXS interface contains the coder-decoder (codec), which converts the spoken analog voice wave into a digital format for processing by the voice-enabled device. This figure depicts an FXO interface. The FXO...

Example Dial Plan Implementations

The North American telephone network is designed around a 10-digit dial plan that consists of 3-digit area codes and 7-digit telephone numbers. For telephone numbers that are located within an area code, the PSTN uses a 7-digit dial plan. Features within a CO-based PBX, such as Centrex, allow the use of a custom 5-digit dial plan for customers who subscribe to that service. PBXs are more flexible and allow for variable-length dial plans containing 3 to 11 digits. Dial plans contain specific...

Specifies the DN of the MoH source

The dial peer that is used to connect the physical E& M or FXO port to the destination pattern that will be used to connect to the MoH feed is created with the dial-peer voice tag pots command. The physical voice port that will be used is associated with the port command and the telephone number used is defined by the destination-pattern command. Router(config) dial peer voice 7777 pots Router(config-dial-peer) port 1 1 0 Associates the dial peer with a voice port Router(config-dial-peer)...

Matching Outbound Dial Peers

Destination pattern is matched based on longest number match Dial-peer voice 1 voip Destination-pattern .T Session target ipv4 10.1.1.1 Dial-peer voice 2 voip Destination-pattern 555 2-3 , Session target ipv4 10.2.2.2 Dial-peer voice 3 voip Destination-pattern 5551 , Session target ipv4 10.3.3.3 Dial-peer voice 4 voip Destination-pattern S5S1234 Session target ipv4 10.4.4.4 Example 1 dialed number 555-1234 will match dial peer 4 Example 2 dialed number 555-1235 will match dial peer 3 Example 3...

Example

The figure illustrates the proper VoIP dial peer configuration on a Cisco voice-enabled router. The dial-peer voice 2 voip command notifies the router that dial peer 2 is a VoIP dial peer with a tag of 2. The destination-pattern 8888 command notifies the router that this dial peer defines an IP voice path across the network for telephone number 8888. The session target ipv4 10.18.0.1 command defines the IP address of the router that is connected to the remote telephony device. This topic...

Example Voice Port Tuning

This example shows voice port tuning parameters on the E& M and FXO ports of a Cisco voice-enabled router. In the example, the PBX output is -4 dB, whereas the voice router functions best at -3 dB. Therefore, the adjustment is made in the inbound path to the router using the input-gain command. The impedance setting on the router needs to be changed from the default of 600r to match the 900c impedance setting for the PBX. Because this is an E& M port, echo cancellation is disabled. The...

Causecode isdnreject value

The CAC for the H.323 VoIP gateways feature allows you to configure thresholds for local resources, memory, and CPU resources. With the call threshold command, you can configure two thresholds, high and low, for each resource. Call treatment is triggered when the current value of a resource exceeds the configured high. The call treatment remains in effect until the current resource value falls below the configured low. Having high and low thresholds prevents call admission flapping and provides...

Example Using Digit Manipulation Tools

The following is a sample configuration using the prefix command dial-peer voice 1 pots destination-pattern 555 prefix 555 port 1 0 0 In the sample configuration using the prefix command, the device attached to port 1 0 0 needs all seven digits to process the call. On a POTS dial peer, only wildcard-matched digits are forwarded by default. Use the prefix command to send the prefix numbers 555 before forwarding the four wildcard-matched digits. The following is a sample configuration using the...

Example Define the COR lists

CMERouter(config) dial-peer list callLocal CMERouter(config-dp-corlist)member local_call CMERouter(config) dial-peer list call911 CMERouter(config-dp-corlist)member 911 CMERouter(config) dial-peer list call1800 CMERouter(config-dp-corlist)member 1800 CMERouter(config) dial-peer list call1900 CMERouter(config-dp-corlist)member 1900 This is the third step to configure Class of Restriction (COR). Steps to Configure Class of Restriction Step 3 - Assign the COR list to the dial peers dial-peer voice...

Example Service Provider Requirements

An IP telephony service provider needs to upgrade their existing gateway platforms because of business growth. The service provider sells a managed IP telephony service to small and medium businesses and provides connections to many different low-cost, long-distance carriers for their customers. Their issues are call quality over the IP network, so delay and jitter need to be controlled. Service providers also must consider scalability and the ability to provide differentiated levels of service...

Cisco Call Manager Express Files Cont Bundled Files

Tar file with all CME 3.1 files for IOS 12.3(7 T zip file with all CME 3.1 files for IOS 12.3f7> T A bundled file with all of the Cisco CME files can be downloaded from cisco.com. The Cisco CME bundle comes in either a tar file or a zip file. These files can then be extracted on the FTP or TFTP server. Reference The Cisco CME software can be found at the following URL Cisco CallManager Express Files (Cont.) Bundled Files The Cisco CME bundle contains all of the files needed to install and...

Must be no existing telephony service configuration

The automated setup is designed for the administrator that does not have a lot of experience with Cisco IOS and may not feel comfortable manually configuring the Cisco CME system. A question-and-answer interface will start and all the administrator does is answer the questions appropriately. Note Any existing configuration of the telephony-service in Cisco CME must be removed prior to Configure NTP prior to running the setup utility Load the firmware files into flash RAM prior to running the...

Timers and Timing

This topic identifies the timing requirements and adjustments that are applicable to voice interfaces. Under normal use, these timers do not need adjusting. In instances where ports are connected to a device that does not properly respond to dialed digits or hookflash, or where the connected device provides automated dialing, these timers can be configured to allow more or less time for a specific function. timing hookflash-in hookflash-out

Two Ephonedns One Number Diff Ephones

Two ephone-dns with one number on different ephones configuration example CMERouter(config) ephone-dn 5 dual line CMERouter(config-ephone-dn) number 1004 CMERouter(config-ephone-dn) preference 0 CMERouter(config-ephone-dn) no huntstop CMERouter(config) ephone-dn 6 dual line CMERouter(config-ephone-dn) number 1004 CMERouter(config-ephone-dn) preference 1 CMERouter(config-ephone-dn) huntstop CMERouter(config) ephone 4 CMERouter(config-ephone) mac-address 000F.2470.F131 CMERouter(config-ephone)...

Loop Start Signaling

Idle State On Hook) Telephone has open 2-wire loop. CO or FX module has battery on ring, ground on tip. Line Seizure (Off Hook) Telephone closes 2-wire loop. CO or FXS module will return dial tone. CO Seizure CO applies AC ring voltage, superimposed over the -48 VDC, Line Seizure (Off Hook) Telephone closes 2-wire loop. Loop-start signaling allows a user or the telephone company to seize a line or trunk when a subscriber is initiating a call. It is primarily used on local loops rather than on...

T1 Interface

All rights re This figure depicts a T1 interface. In a corporate environment with a large volume of voice traffic, connections to the PSTN and to PBXs are primarily digital. A T1 interface is a form of digital connection that can simultaneously carry up to 24 conversations using two-wire pairs. When a T1 link operates in full-duplex mode, one wire pair sends and the other wire pair receives. The 24 channels are grouped together to form a frame. The frames are then...

Mean Opinion Score

The figure depicts mean opinion score (MOS). MOS is a system of grading the voice quality of telephone connections. The MOS is a statistical measurement of voice quality derived from the judgments of several subscribers. Graded by humans and very subjective, the range of MOS is 1 to 5, where 5 is direct conversation. Voice Quality of Telephone Connections

Enterprise Gateway Considerations Remote Site

As IP telephony services become a standard in the corporate environment, a broad mix of requirements surface in the enterprise environment. The IP telephony deployment typically begins by connecting to the PSTN to manage off-net calls and using a Cisco CallManager infrastructure to manage on-net calls.

Centralized Call Control vs Distributed Call Control

Ease of dial plan consistency and updating Dial plan consistency and updating is more difficult Supplementary services (PBX features) Supplementary services harder to implement Difficult to scale all new features and applications must be implemented on the central controller, central breakpoint, or bottleneck Scalable need more applications functions or performance. Add more servers and they can be located anywhere Difficult to provide resiliency over network failures Difficult to add new...

Example Gateway Interconnect Considerations

The table shows examples of questions that you must ask to determine the requirements for gateway interconnections. Determining Gateway Interconnection Requirements You must ensure support for proper call processing, such as Media Gateway Control Protocol (MGCP), session initiation protocol (SIP), or H.323. Distributed call processing is easier to implement, but costs are higher when deploying intelligent devices at each site. Is remote site survivability an issue Remote site survivability is...

CMERouter Configephonedn 2 dualline CMERouter Configephonednnumber 1002

The ephone-dn creates one virtual voice port The dual-line keyword indicates two voice channels for calls to terminate on an ephone-dn extension Use on ephone-dns that need call waiting, consultative transfer, or conferencing on one button Cannot be used on ephone-dns used for intercoms, paging, MWI or MoH feeds 2005 Cisco Systems, Inc. All rights re A dual-line ephone-dn has the following characteristics Can make two call connections at the same time using one phone line button. A dual-line...

How Does Cisco Call Manager Express Work

This topic describes how Cisco CME system works. The Cisco CME system provides the PBX-like features and functions for the IP phones. These features are a result of the concept of a centralized point of control and intelligence. The Cisco CME router provides all of the call control and intelligence needed for the IP phones to place and receive calls. In a Cisco CME deployment, the IP phones are not capable of setting up a call by themselves. In fact, the IP phones are totally under the control...

Connection tieline

Emulates a temporary tie-line trunk to a PBX You can configure voice ports to support special connection requirements. These requirements usually reflect the needs of a specific business environment that must connect to the network in a special way. The following is a list of available connection commands and their application connection plar Private line, automatic ringdown (PLAR) is an autodialing mechanism that permanently associates a voice port with a far-end voice port, allowing call...

Bandwidth Implications of Codec

One of the most important factors for the network administrator to consider while building voice networks is proper capacity planning. Network administrators must understand how much bandwidth is used for each VoIP call. With a thorough understanding of VoIP bandwidth, the network administrator can apply capacity-planning tools. Following is a list of codecs and their associated bandwidth G.711 The G.711 pulse code modulation (PCM) coding scheme uses the most bandwidth. It takes samples 8000...

Cisco Call Manager Express Files Cont TAPI Integration

To allow a third-party piece of software to interact with the Cisco CME system through TAPI lite, the files in the IOS TSP file will need to be installed on the same Windows PC where the software will be installed. This will be covered in more detail in a later module. Ef ijnst32i.ex_ j Q_ISDel.exe _Setup.dll _sysi.cab _sysl.hdr C _userl.cab _userl.hdr IJciscoIOSTSP.tsp CiscoIOSTUISP.dll B Data, tag datai.cab datal.hdr lang.dat layout, bin LogTrace.dll los.dat Qremovetsp.exe Q setup.bmp...

Productivity Tools

This topic describes the Cisco CME productivity tools. Certain PSTN services, such as three-way calling and call waiting, require hookflash intervention from a phone user. A new soft key labeled Flash has been introduced to provide this functionality on FXO lines attached to the Cisco CME system. The Flash soft key is enabled using the fxo hook-flash command. Once Flash has been enabled and a reboot of the IP phone performed, the softkey is available to provide hookflash functionality during...

On different ephones

Used when two different ephones need the same number A call on hold can be retrieved only by the ephone that put the call on hold There are two different ways for multiple ephone-dns with the same extension number to be utilized. One way is for multiple ephone-dns to be assigned to the same ephone but on separate line buttons. This type of configuration will be useful when more than two calls at a time may arrive at a destination and need to be handled. For example, if 6 calls at a time need to...

VoIP Protocols and the OSI Model

S oft phone Call Manager Human Speech Frame Relay (FR), ATM, Ethernet, Multilink Point-to-Point Protocol (MLPPP), Pornt-to-Point Protocol (PPP), High-Level Data Link Control (HDLC) Constant Voice media packets use RTP UDP Variable Several signaling methods and link layer protocols Constant Voice media packets use RTP UDP Variable Several signaling methods and link layer protocols

Authentication is part of the protocol

H.323 is a specification for transmitting audio, video, and data across an IP network, including the Internet. H.323 is an extension of the ITU Telecommunication Standardization Sector standard H.320. Tip The ATA will need to be configured with H.323 when fax machines are connected to the 2005 Cisco Systems, Inc. All rights reserved. 2005 Cisco Systems, Inc. All rights reserved. H.323 is a specification for transmitting audio, video, and data across an IP network, including the Internet. H.323...

Number of Ephonedns

This topic explains the number of ephone-dn. This topic explains the number of ephone-dn. router(config-telephone) max-dn max-dn Sets the maximum definable number of ephone-dns that may be configured in the system The maximum number of ephone-dns supported is a function of the license and hardware platform The maximum number of ephone-dns that can be configured is based upon the hardware platform that the Cisco CME software is installed on. The default of a newly installed Cisco CME system is...

Digit Manipulation Commands

Adds digits to the front of the dial string before it is forwarded to the telephony interface Controls the number of digits forwarded to the telephony interface Expands an extension into a full telephone number or replaces one number with another Digit translation rules are used to manipulate the calling number, or ANI, or the called number, or DNIS, digits for a voice call Digit manipulation is the task of adding or subtracting digits from the original dialed number to accommodate user-dialing...

Configure System Administrator Credentials

The Cisco CME GUI uses HTTP to transfer information from the Cisco CME router to the PC of an administrator or phone user. The router must be configured as an HTTP server and have the proper Web files in flash locally to serve up to the browser. In addition an initial system administrator username and password must be defined from the router command-line interface (CLI). Customer administrators and phone users can be added from the Cisco CME router using CLI commands or from a PC using GUI Web...

Optional Parameters Locale Parameters

This topic identifies a router configuration. This topic identifies a router configuration. Optional Parameters Locale Parameters Specifies the language for display on an IP phone Specifies the set of call progress tones and cadence On the Cisco IP Phone 7940 and Cisco IP Phone 7960, the language displayed on the phone and the locale for call progress tones and cadences can be set to one of several ISO-3166 codes that indicate specific languages and geographic regions. Note The 7920 IP phone...

DHCP Service Setup Cont Phone Bootup

Free Xml For Cisco Phone

The phone performs a Power on Self Test (POST) Through CDP the IP phone learns what the auxiliary VLAN is On the Cisco CME router a DHCP Scope can be configured. The scope should define the following Range of available IP addresses The address of the TFTP server 2005 Cisco Systems, Inc. All rights re When an IP phone first receives power, the following steps happen Receives power - The IP phone receives power POST - The phone will perform some basic tests called a Power On Self Test (POST)....

Cisco Call Manager Express Files Cont GUI Files

One of the individual files that can be downloaded is the tar that contains the GUI Web interface for Cisco CME. The Cisco Unity Express module GUI is also dependent on the CallManager GUI. Cisco CallManager Express Files (Cont.) GUI Files The contents of the GUI Web interface tar file are shown above and these files will need to be present in the flash of the Cisco CME router. This will be covered in more detail in a later module.

Example Apply the COR to the dial peer

CMERouter(config) dial-peer voice 1 pots 1500 CMERouter(config-dial-peer) port 1 0 0 CMERouter(config-dial-peer) corlist incoming call911 CMERouter(config) dial-peer voice 2pots 1800 CMERouter(config-dial-peer) port 2 1 CMERouter(config-dial-peer) corlist outgoing call1800 Steps to Configure Class of Restriction Step 4 - Assign the COR list to the ephone-dns Defines an ephone-dn and enters ephone-dn mode Specifies a COR list to be used when the ephone-dn is used as either the incoming or...

Channel Associated Signaling CAS

This topic describes the commands required to configure a Channel Associated Signaling (CAS) interface. Basic T1 E1 Controller Configuration Configures the linecode for a T1 line Configures the linecode for a E1 line Use the linecode command to identify the physical layer signaling method to satisfy the ones density requirement on the digital facility of the provider. Without a sufficient number of ones in the digital bit stream, the switches and multiplexers in a WAN can lose their...

Symptoms of Jitter on a Network

This topic provides examples of output for the show call active voice command, which can be used to determine the size of jitter problems. ConnectionId 0xECDE2E7B 0xF46A003F 0x0 0x47070A4 IncomingConnectionId 0xECDE2E7B 0xF46A003F 0x0 0x47070A4 The figure shows sample output for the show call active voice command. Several fields in the show call active voice command output that can help you to determine the actual size of the ReceiveDelay The playout delay for jitter compensation plus the...

Example Acceptable Delay

The G.114 recommendation is for one-way delay only and does not account for round-trip delay. Network design engineers must consider all delays, variable and fixed. Variable delays include queuing and network delays, while fixed delays include coder, packetization, serialization, and dejitter buffer delays. The table is an example of calculating delay budget.

Common Channel Signaling CCS Overview of Isdn Bri Configuration Commands

Configuring ISDN BRI requires global and interface configuration commands. This topic provides an overview of configuration commands required to successfully configure an ISDN BRI connection. At the global level, the administrator must specify the ISDN service provider CO switch type. There are several types of switches to choose from and some of these require special parameters. Standards signaling specifics differ by region. Therefore, the switch type varies according to its geographical...

Common Channel Signaling CCS Overview of Isdn Pri Configuration Commands

Pri Signaling

Configuring ISDN PRI requires global and interface configuration commands. Selecting the correct switch type to which to connect is crucial when configuring ISDN PRI. This topic provides an overview of the isdn switch-type command. Use the isdn switch-type command to specify the central office PRI switch to which the router connects. With Cisco IOS Release 11.3(3)T or later, this command is also available as a controller command to allow different switch types to be supported on different...

Impact of Voice Samples

Voice sample size is a variable that can affect total bandwidth used. A voice sample is defined as the digital output from a codec DSP that is encapsulated into a protocol data unit (PDU). Cisco uses DSPs that output samples based on digitization of 10 ms-worth of audio. Cisco voice equipment encapsulates 20 ms of audio in each PDU by default, regardless of the codec used. You can apply an optional configuration command to the dial peer to vary the number of samples encapsulated. When you...

Configuring a transfer pattern

Call transfer is a function that can be configured in various ways, depending on the supported protocols. These call transfer commands include system-wide settings that can be overridden with phone specific settings, which can be overridden by settings on the transfer pattern. transfer-system blind full-blind full-consult local-consult Specifies the call transfer method for all Cisco CME extensions full-blind To specify the call transfer method for IP phone extensions system-wide that use the...

Example Show Auto QoS and Show Auto QoS Interface

The show auto qos command displays all of the configurations created by the AutoQoS VoIP feature. interface Serial6 1.1 point-to-point frame-relay interface-dlci 100 class AutoQoS-VoIP-FR-Serial6 1-100 frame-relay ip rtp header-compression map-class frame-relay AutoQoS-VoIP-FR-Serial6 1-100 service-policy output AutoQoS-Policy-UnTrust frame-relay fragment 640 show policy-map interface interface type Displays the packet statistics of all classes that are configured for all service policies...

Robbed Bit Signaling

X Least significant bit in each DSO is robbed for signaling every sixth frame Because each DS0 channel carries 64 kbps, and G.711 is 64 kbps, there is no room to carry signaling. Implemented for voice, the T1 uses every sixth frame to convey signaling information. In every sixth frame, the least significant bit (LSB) for each of the voice channels is used to convey the signaling. Although this implementation detracts from the overall voice quality (because only seven bits represent a sample for...

Major VoIP Protocols

ITU standard protocol for interactive conferencing. Evolved from H.320 ISDN standard. Flexible, complex. Force (IETF) standard for PSTN gateway control, thin device control. IETF protocol for interactive and n on interactive conferencing. Simpler, but less mature, than H.323. IETF standard media streaming protocol. IETF protocol that provides out-of-band control information for an RTP flow. The major VoIP protocols include the following H.323 An ITU standard protocol for interactive...

Configuring AutoQoS Cisco Catalyst 6500 Switch

Global configuration command All the global QoS settings are applied to all ports in the switch Prompt displays showing the CLI for the port-based automatic QoS commands currently supported Console> (enable)set qos autoqos QoS is enabled All ingress and egress QoS scheduling parameters configured on all ports.CoS to DSCP, DSCP to COS, IP Precedence to DSCP and policed dscp maps configured. Global QoS configured, port specific autoqos recommended set port qos < mod port> autoqos trust <...

Physical Connectivity Options

This figure depicts physical connection options for IP Phones. The IP Phone connects to the network through a Category 5 or better cable that has RJ-45 connectors. The power-enabled switch port or an external power supply provides power to an IP Phone. The IP Phone functions like other IP-capable devices sending IP packets to the IP network. Because these packets are carrying voice, you must consider both logical and physical configuration issues. At the physical connection level, there are...

Individual Cisco CME Files Firmware files

CP79050101SCCP030530B31.ZUO 790S ZLip file Unsigned signed phoneload for7960 40 10 use .bin for unsigned to signed load upgrading, use .sbin for signed load upgrading, tar file for CME 3.1 system w 12.3(7)Tw fixfor CSCed73192 tar tile for CME 3.1 GUI files w 123(7)1 w fix for CSCei73192 The files can be downloaded individually as well as in a bundle. Note These files are version-specific and are not backwards compatible

Live audio source via a feed

MoH is an audio stream that is played to PSTN and VoIP G.711 callers who are placed on hold by phones in a Cisco CME system. This audio stream is intended to reassure callers that they are still connected to their calls. MoH is not played to local Cisco CME phones that are on hold with other Cisco CME phones. These parties hear a periodic repeating tone instead. The audio stream that is used for MoH can derive from one of two sources an audio file or a live feed. If both are configured...

Effects of VAD on Bandwidth

This topic describes the effect of voice activity detection (VAD) on total bandwidth. On average, an aggregate of 24 calls or more may contain 35 percent silence. With traditional telephony voice networks, all voice calls use 64-kbps fixed-bandwidth links regardless of how much of the conversation is speech and how much is silence. In Cisco VoIP networks, all conversations and silences are packetized. VAD suppresses packets of silence. Instead of sending VoIP packets of silence, VoIP gateways...

Informational Signaling with Call Progress Indicators

Call-progress indicators in the form of tone combinations are used to notify subscribers of call status. Each combination of tones represents a different event in the call process, as follows Dial tone Indicates that the telephone company is ready to receive digits from the user telephone. The Cisco routers provide dial tone as a method of showing that the hardware is installed. In a PBX or key telephone system, the dial tone indicates that the system is ready to receive digits. Busy tone...

Rebooting Cisco Call Manager Express Phones

This topic discusses rebooting IP phones. This topic discusses rebooting IP phones. Rebooting Cisco CallManager Express Phones After you update information for one or more phones associated with a Cisco CME router, the phone or phones must be rebooted. There are two commands to reboot the phones reset and restart. The reset command performs a hard reboot similar to a power-off-power-on sequence. It reboots the phone and contacts the DHCP server and TFTP server to update from their information...

Message

This graphic shows the different areas that will be present on the display of a Cisco CME controlled phone. Theses features can be customized for the current implementation. Enters ephone-dn configuration mode CMERouter(config-ephone-dn) description display-text Enters the header bar for the IP Phone CMERouter(config-ephone-dn) label string Configures a label on the line instead of the line number The description command is use to change the IP header bar of a phone. A common use of this...