ACTIV! IP - CHANNEL MANAGER CONFIGURATION …_IP_Channel_Manager.pdf · 4 Activ! IP - Channel Manager (rev D) Introduction Activ! IP’s recording solutions are compatible and integrate

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Activ! IP - Channel Manager (rev D)

VPI • 160 Camino Ruiz, Camarillo, CA 93012-6700(Voice) 800-200-5430 • 805-389-5200 • (Fax) 805-389-5202

www.vpi-corp.com

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TOC

Contents

Introduction ------------------------------------------------------------------------------------------------------------------------------------ 4Activ! IP Recording Solution Highlights ............................................................................................................... 4

Terminology ------------------------------------------------------------------------------------------------------------------------------------ 5AudioCodes and SMARTWORKS® API’s .......................................................................................................... 5CMAPI ------------------------------------------------------------------------------------------------------------------------------------- 5Gateway ---------------------------------------------------------------------------------------------------------------------------------- 5Jitter ---------------------------------------------------------------------------------------------------------------------------------------- 5Packet ------------------------------------------------------------------------------------------------------------------------------------- 5Packet Sniffer ---------------------------------------------------------------------------------------------------------------------------- 5Port Mirroring ---------------------------------------------------------------------------------------------------------------------------- 6RTP ---------------------------------------------------------------------------------------------------------------------------------------- 6SIP (Session Initiation Protocol) ----------------------------------------------------------------------------------------------------- 6SPAN Port -------------------------------------------------------------------------------------------------------------------------------- 6Brief Description of SPAN ------------------------------------------------------------------------------------------------------------- 7Switch and Port Monitoring ----------------------------------------------------------------------------------------------------------- 7Tap Environment ------------------------------------------------------------------------------------------------------------------------- 7TCP ---------------------------------------------------------------------------------------------------------------------------------------- 7TDM (Time Division Multiplexing) ---------------------------------------------------------------------------------------------------- 8UDP ---------------------------------------------------------------------------------------------------------------------------------------- 8Upstream/Downstream Traffic -------------------------------------------------------------------------------------------------------- 8Virtual LAN ------------------------------------------------------------------------------------------------------------------------------- 8VoIP ---------------------------------------------------------------------------------------------------------------------------------------- 9VoIP Phone------------------------------------------------------------------------------------------------------------------------------- 9

Hardware Overview --------------------------------------------------------------------------------------------------------------------------10SmartWORKS® IPX (Event Monitor) ----------------------------------------------------------------------------------------------10

Example of Incoming Traffic ---------------------------------------------------------------------------------------------------- 11Decoding Logic -------------------------------------------------------------------------------------------------------------------- 11D-Channel Events ----------------------------------------------------------------------------------------------------------------12PBX Command Events ----------------------------------------------------------------------------------------------------------12Phone Action Events -------------------------------------------------------------------------------------------------------------12

Software Configuration (VPConfig) VoIP Channel Manager ..................................................................................... 13Operational Overview ------------------------------------------------------------------------------------------------------------------13SPAN Overview -------------------------------------------------------------------------------------------------------------------------13SPAN Tap Point Options -------------------------------------------------------------------------------------------------------------14IPX Setup: VPConfig > Channel Manager > IPX (Tab) > Settings (Tab) ............................................................ 15

IPX Setup ---------------------------------------------------------------------------------------------------------------------------15Ignore Extensions/IP -------------------------------------------------------------------------------------------------------------17Record Extensions/IPs ----------------------------------------------------------------------------------------------------------17Called and Calling Party Transformations .................................................................................................. 17Party Transformation - Cisco Wildcards .................................................................................................... 19Apply/Cancel Buttons -----------------------------------------------------------------------------------------------------------20

IPX Setup: VPConfig > Channel Manager > IPX (Tab) > Boards (Tab) .............................................................. 21IPX Boards -------------------------------------------------------------------------------------------------------------------------21Signaling Protocols --------------------------------------------------------------------------------------------------------------21Board Protocol (i.e., Cisco®) --------------------------------------------------------------------------------------------------25Apply/Cancel Buttons -----------------------------------------------------------------------------------------------------------25

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© Copyright 2007 VPI, Inc.

The information contained herein is subject to change without notice.

As the registered end user, you are free to make and use as manycopies of this manual as you like, so long as it is used for internalpurposes only and is not transmitted in any manner whatsoever to anyindividual, organization or corporate entity not specifically named inyour End User License Agreement.

VPI shall not be liable for technical or editorial errors or omissionscontained herein.

VPConfig > Channel Manager > Database (Tab) .............................................................................................. 26Database Mappings --------------------------------------------------------------------------------------------------------------26Database Field --------------------------------------------------------------------------------------------------------------------27Apply/Cancel Buttons -----------------------------------------------------------------------------------------------------------27

VPConfig > Channel Manager > Software RTP (Tab) ........................................................................................ 28RTP Setup -------------------------------------------------------------------------------------------------------------------------28Destination IP Address ----------------------------------------------------------------------------------------------------------28Time Intervals ----------------------------------------------------------------------------------------------------------------------29Ports---------------------------------------------------------------------------------------------------------------------------------29RTP Codec -------------------------------------------------------------------------------------------------------------------------29Apply/Cancel Buttons -----------------------------------------------------------------------------------------------------------29

Skinny Protocol Database Fields --------------------------------------------------------------------------------------------------------30Cisco® Skinny Redirect Reasons -------------------------------------------------------------------------------------------------------31Cisco® Specific Features Supported ---------------------------------------------------------------------------------------------------32Activ! IP Known Technical Limitations --------------------------------------------------------------------------------------------------37Index -------------------------------------------------------------------------------------------------------------------------------------------39Revision History -----------------------------------------------------------------------------------------------------------------------------41

TOC

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Introduction

Activ! IP’s recording solutions are compatible and integrate with industry leading VoIP Telephony platforms whichinclude Cisco® Systems, Avaya® IP/Office IP, Nortel® Meridian 1 and the BCM® IP PBX, Ericson®, Siemens®, Alcatel®,Asterisk® and SIP . Through collaboration with major Voice Over IP telephony solution developers, VPI ensures that itsclients are able to effectively leverage VoIP recording solutions to capture, evaluate, analyze and improve multimediainteractions over converging networks, supporting transition from legacy PBX systems to the new ROI boosting VoiceOver IP telephony-based solutions.

Activ! IP Recording Solution Highlights

• Ready migration path to Voice Over IP telephony to gain total cost of ownership savings due to less expensivecalling rates

• Combine VoIP and circuit-switched telephony networks, preserving your organization’s investment in traditionalACDs/PBXs

• Integration with VPI’s Activ! Suite of products for customer experience management, including Activ! ViewDesktop Screen Capture and Activ! IQ Call Center Quality Monitoring technologies

• Activ! IP supports G.711and G.729a codecs.

• The combined solution delivers special features to include:

• Keep/delete calls as they are recorded• Combine on-demand start and stop VoIP recording with Activ IP’s full-time IP Recorder• Custom applications

• Completely D-Channel event driven solution that requires no third party software and/or connection to the PBXin order to work. The communication between the PBX and the phone provide the start and stop events for eachphone. Activ IP uses and captures all possible data provided in this data stream. Some PBX’s provide moredata than others.

• Hybrid System-Activ! IP will support both analog VOX recording and D-channel driven VoIP recording in onesystem.

• Easily upgrade existing systems with VoIP recording channels.

Introduction

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Terminology

This section explains the pertinent terms relating to Activ! IP’s capturing and recording capabilities in the VoIP environment(in alphabetical order).

AudioCodes and SMARTWORKS® API’s

AudioCodes and SmartWorks API’s (Application Program Interfaces) provide the set of routines, protocols, andtools for building software applications to communicate with Activ! IP and AudioCodes recording solution products.

CMAPI

Avaya CMAPI (Communication Manager Application Program Interface).

Gateway

A gateway is either hardware or software that acts as a bridge between two networks so that data can betransferred between a number of computers. For example, when you send an e-mail or when you log into a Website, there is a gateway that allows the connection to take place. Often, your connection to a Web site will involvemany smaller connections to other servers along the way. In these cases, a number of gateways are used.

Jitter

In voice over IP (VoIP), jitter is the variation in the time between packets arriving, caused by network congestion,timing drift, or route changes. A jitter buffer can be used to handle jitter.

Packet

A packet is a formatted block of information carried by a computer network. Computer communications links thatdo not support packets, such as traditional point-to-point telecommunications links, simply transmit data as aseries of bytes, characters, or bits alone. When data is formatted into a packet, the network can transmit longermessages more efficiently and reliably.

Packet Sniffer

A Packet sniffer (also known as network or protocol analyzer or Ethernet sniffer) is computer software (usually) orcomputer hardware that can intercept and log traffic passing over a digital network or part of a network. As datastreams travel back and forth over the network, the sniffer captures each packet and eventually decodes andanalyzes its content according to the appropriate RFC (Request for Comments) or other specifications. Dependingon the network structure (hub or switch) one can sniff all or just parts of the traffic narrowing by switches to gainaccess to traffic from other systems on the network (i.e., ARP spoofing). For network monitoring purposes it mayalso be desirable to monitor all data packets in a LAN by using a network switch with a so-called monitoring port,whose purpose is to mirror all packets passing through all ports of the switch.

Terminology

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Port Mirroring

Port mirroring is used on a network switch to send a copy of all network packets seen on one switch port to anetwork monitoring connection on another switch port. This is commonly used for network appliances thatrequire monitoring of network traffic, such as an intrusion-detection system. Port mirroring on a Cisco Systemsswitch is generally referred to as Switched Port Analyzer (SPAN).

RTP

RTP (Real Time Protocol) is a packet based communication protocol that adds timing and sequence informationto each packet to allow the reassembly of packets to reproduce real time audio and video information. RTP is atransport used in IP audio and video environments.

SIP (Session Initiation Protocol)

A protocol designed to initiate, maintain and terminate a session involving multimedia streams such as voice,data and video. All SIP messages are either a request or a response to a request.

SPAN Port

Another method to monitor networks is to use port mirroring is called "SPAN" (Switched Port Analyzer) by Cisco,and given other names by some other vendors on routers and switches (i.e., PSPAN, VSPAN, LSPAN) for moreinformation refer to the “SPAN Overview” section on page 15. This is a low-cost alternative to network taps, andsolves many of the same problems.

Terminology

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Brief Description of SPAN:

What is SPAN and why is it needed? The SPAN feature was introduced on switches because of a fundamentaldifference that switches have with hubs. When a hub receives a packet on one port, the hub sends out acopy of that packet on all ports except on the one where the hub received the packet. After a switch boots,it starts to build up a Layer 2 forwarding table on the basis of the source MAC address of the differentpackets that the switch receives. After this forwarding table is built, the switch forwards traffic that isdestined for a MAC address directly to the corresponding port.

In the above diagram, the sniffer is attached to a port that is configured to receive a copy of every packetthat host A sends. This port is called a SPAN port.

Switch and Port Monitoring

Switch is a network exchange facility operating at the data link layer (layer 2) and sometimes the network layer(layer 3) of the OSI Reference Model.

Unlike hubs, switches prevent promiscuous sniffing. When a data packet is transmitted in non-promiscuousmode, all the LAN devices "listen to" the data to determine if the network address included in the data packet istheirs. If it isn't, the data packet is passed onto the next LAN device until the device with the correct networkaddress is reached. That device then receives and reads the data.However, most modern switches (management switches) support "port mirroring", which is a feature that allowsyou to configure the switch to redirect the traffic that occurs on some or all ports to a designated monitoring porton the switch. With this feature, you can monitor the entire LAN segment in switched network environment.

Tap Environment

For IP call recording applications, active interfaces can be connected directly to an available mirror port or can beconnected passively anywhere within the IP-PBX configuration.

TCP

Abbreviation of Transmission Control Protocol, and pronounced as separate letters. TCP is one of the mainprotocols in TCP/IP networks. Whereas the IP protocol deals only with packets, TCP enables two hosts toestablish a connection and exchange streams of data. TCP guarantees delivery of data and also guarantees thatpackets will be delivered in the same order in which they were sent.

Terminology

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TDM (Time Division Multiplexing)

The means by which many different signals can be carried on the same transmission path by interleaving theseportions of signals over time. This means that circuit switched networks can be used more efficiently. Thistechnology is used on traditional networks such as the PSTN and GSM mobile networks. The main difference topacket switching is that the time slots are pre-allocated to the channels rather than being allocated on a per-timeslot basis.

UDP

Abbreviation for User Datagram Protocol, a connectionless protocol that, like TCP, runs on top of IP networks.Unlike TCP/IP, UDP/IP provides very few error recovery services, offering instead a direct way to send and receivedatagrams over an IP network. It's used primarily for broadcasting messages over a network.

Upstream/Downstream Traffic

Upstream network traffic flows away from the local computer toward the remote destination. Conversely, downstreamtraffic flows to the user's computer. Traffic on most networks flows in both upstream and downstream directionssimultaneously, and often when data flows in one direction, network protocols often send control instructions(generally invisible to the user) in the opposite direction.

One way to generate upstream traffic is to upload files to a server or send an email message. Conversely,downloading files and receiving email generate downstream traffic. Typical Internet users create much moredownstream than upstream traffic.

Examples: The Web browser sends HTTP requests upstream to the Web server, and the server replies withdownstream data usually in the form of HTML pages.

• Ingress (RX) - Packets/Frames entering the device or network being described. In data/center orserver centric discussions sometimes referred to as client side or client traffic. That is traffictransmitted by the client and received by the data center.

• Egress (TX) - Packets/Frames leaving the local device or network. In data/center or server centricdiscussion it is traffic transmitted by the device or network and received by the client.

Virtual LAN

A virtual LAN, commonly known as a vLAN or as a VLAN, is a method of creating independent logical networkswithin a physical network. Several VLANs can co exist within such a network. This helps in reducing the broadcastdomain. A VLAN consists of a network of computers that behave as if connected to the same wire - even thoughthey may actually be physically connected to different segments of a LAN. Network administrators configureVLANs through software rather than hardware, which makes them extremely flexible. One of the biggest advantagesof VLANs emerges when physically moving a computer to another location: it can stay on the same VLANwithout the need for any hardware reconfiguration.

Terminology

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For Activ! IP the voice traffic is isolated onto a separate VLAN on each of the ports that are connected to a phone.The switch port that is configured for connecting a phone would have separate VLANs that are configured forcarrying the following:

• Data traffic to and from the PC that is connected to the switch through the access port of the IP phone(i.e., VLAN 1)

• Voice traffic to and from the IP phone (i.e.,VLAN 2)

Isolating the phones on a separate, auxiliary VLAN increases the quality of the voice traffic and allows a largenumber of phones to be added to an existing network where there are not enough IP addresses.

Advantages of VLAN:

• Reduces the broadcast domain, which in turn reduces network traffic and increases networksecurity (both of which are hampered in case of single large broadcast domain)

• Reduces management effort to create subnetworks

• Reduces hardware requirement, as networks can be logically instead of physically separated

• Increases control over multiple traffic types

• Increases network security

VoIP

Voice Over IP is the technology that provides the capability to break voice streams down into small pieces, groupthem together in an IP Packet, and then send them over an IP network to the far end caller.

VoIP Phone

A VoIP phone is a telephone device that looks like a traditional telephone. However, instead of connecting to thePOTS (plain old telephone system) network, it instead has an Ethernet port that is used to connect to a TCP/IPcomputer network.

Terminology

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Hardware Overview

SmartWORKS® IPX (Event Monitor)

SmartWORKS® IPX monitors and collects voice packets in the IP-PBX environment and reports D-Channel datato Activ! IP via the SmartWORKS® API. Supporting multiple protocols, it has on-board connections for monitoringup to 64 channels (120 in the future) of upstream and downstream Real Time Protocol (RTP) and one on-boardconnection for output to the Recording Processing Board (i.e., Standard Network Card).

The IPX does not have any DSP resources. This board is not designed for signal detection or recording purposes.The IPX has been designed with packet forwarding capabilities which enable the user to forward media packetsto another system on the network for signal processing and recording.

• Provides the Tap Point• Monitors PBX Events• Multiple IPX per server• Limited to Half Ethernet bandwidth

VoIP Tap Methods:

• Mirror/span port (Most likely method for station-side tap).• Tap box (Most likely method for trunk-side tap).

Signaling Protocol—currently supported (more details in the VPCONFIG-VoIP Mgr. Config Section):

• Cisco Skinny (SCCP Protocol-default)• Avaya H323 (IP PBX S8300 or IP Office PBX)• H323 Protocol (Standard)• Nortel UNIStim (Meridian or BCM PBX)• Ericsson H323 (MD110 PBX)• SIP (Standard)• Siemens (HiPath 4000 PBX)• Alcatel (OMNI PCX)

Note: Future protocol support accommodated with a software upgrade.

Cisco® Skinny Client Control Protocol (default)

SCCP is a proprietary terminal control protocol originally developed by Selsius Corporation. It is nowowned and defined by Cisco® Systems, Inc. as a messaging set between a skinny client and the Cisco®

CallManager. Examples of skinny clients include the Cisco® 7900 series of IP phone such as the Cisco®

7960, Cisco® 7940 and the 802.11b wireless Cisco® 7920. Skinny is a lightweight protocol which allows forefficient communication with Cisco® Call Manager. Call Manager acts as a signalling proxy for call eventsinitiated over other common protocols such as H.323, SIP, ISDN and/or MGCP.

A skinny client uses TCP/IP to and from one or more Call Managers in a cluster. RTP/UDP/IP is used toand from a similar skinny client or H.323 terminal for the bearer traffic (real-time audio stream). SCCP is astimulus-based protocol and is designed as a communications protocol for hardware endpoints and otherembedded systems, with significant CPU and memory constraints.

Hardware Overview

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Example of Incoming Traffic

To understand how the IPX works, follow this example of incoming traffic:

1.Two Ethernet ports monitor the network- one for upstream (Tx) traffic, and the second for downstream (Rx)traffic. The two sides of the conversation are not summed on the IPX.

2.The IPX sorts packets based on protocol type. As call set-up is negotiated signaling packets are passed tothe appropriate protocol decoding stack.

3.Signaling packets, based on the endpoint’s IP address, are tagged with a Station ID by the IPX’s StationManager.

4.D-Channel messages are decoded and events are passed to the user application via the Event Mailer. Eachevent is reported with the Station ID.

5.Call State information is abstracted and reported to the user application as Call Control events. Station ID isreported with each event.

6.When an RTP connection is established between endpoints the event is reported to the user application.The Session Manager assigns a unique Session ID to this connection which is passed to the user application along with the Station ID.

7.All media packets (RTP) are redirected to the RTP Forwarding Process.

8.RTP packets are forwarded to two ports on the recording device.

9.When the call is disconnected:

The RTP logical channel is closed - the event is reported. The Session manager gives up this Session IDfor re-use by the system.

During call tear down all D-Channel information is decoded and all signaling information is reported to theuser application in the form of D-Channel events. Call Control information is also reported to Activ! IP.

Decoding Logic

The IPX presents four types of line events to Activ! IP:

• D-Channel - signaling and terminal control messages are decoded and passed to Activ! IP

• Call Control Events - an underlying call state machine abstracts information and provides call state eventreporting

• Media Events - all media (RTP) connections are monitored and reported to Activ! IP

• Station Events - alerts the application when VoIP endpoints are added or removed from the monitorednetwork

Hardware Overview

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D-Channel Events

The IPX provides D-Channel decoding similar to the D-Channel decoding on the AudioCodes® NGX boards. Toobtain D-Channel information, the tap must be positioned between the local VoIP phones and the IP-PBX (VoIPCall Agent). Activ! IP enables the protocol decoding stack on the board, and then enables D-Channel eventreporting for this protocol.

The D-Channel decoder provides a message-to-event translation for the terminal control messages. AudioCodes®

groups all D-Channel events into two types:

• PBX Command events - messages initiated by the PBX to control the phone

• Phone Action Events - messages initiated by the phone to inform the PBX of an action taken

PBX Command Events

The following types of command events are reported to Activ! IP:

• Signaling - these events indicate the PBX is commanding the phone to produce a tone (ringing, or incomingpage)

• Audio Events - indicate the PBX is controlling external audio devices such as headsets or microphones

• LEDs - these events correspond to light changes on the phone. Light events are important indications whenmonitoring call states and feature activity

• Display - these events indicate that the LCD on the phone has been updated. These are usually related tothe clock display, or messages displayed on the LCD

• Call State - these events are generated with a change in call state (NOTE: These are not related to Call Controlevents)

Phone Action Events

These events are generated by the phone after an action has been taken (i.e., button pressed). The phone isinforming the PBX that something has occurred. Events generated by the phone have been classified by thefollowing types:

• Hook State - off hook and on hook changes occur when the handset is removed or replaced

• Button Depression events - indicate that a button on the phone was used. For example: digits, speakerbuttons etc. Button events can include both a pressed or released event, depending on the PBX.

Hardware Overview

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Software Configuration (VPConfig) VoIP Channel Manager

Operational Overview

VPI’s VoIP D-Channel recording solution consists of a combination of at least two PCI cards. The first card is the“SmartWorks® IPX” card. The IPX card is a packet sniffing card that connects directly to the SPAN port of the IPSwitch for the phone/devices and supports most of the currently used VoIP PBX’s and signaling protocols. It has atotal of three Ethernet ports on it. The top two, (ports 0 and 1) are the inputs and the bottom port, (port 2) is theoutput which streams only the RTP packets containing the call(s) audio to the recording card(s).The IPX card alsosupplies all of the Call Control and D-Channel events for each call that it detects traffic on. These events are used todetermine when to start and stop recording while also supplying all pertinent call data for each call which is thensaved by Activ! IP to become searchable data. Each IPX card can accept inputs from two individual SPAN portssimultaneously and several IPX cards can be used in one system in conjunction with one recording card.

Activ! IP uses a standard network card to perform the actual recording of the RTP packets received from the IPX card(s)output. All VoIP traffic contains an audio path per party involved in a call (one per party). The recording card must recordeach party of the call un-summed then combine these two un-summed recordings into one WAV file hence creating onerecording per call. Activ! IP currently supports G.711, G.729 and G.723 codecs on a per call basis. Each call is checkedbefore recording starts to determine the codec, port(s) and signaling protocol type of each call automatically.

SPAN Overview

A SPAN session is a feature of the Cisco® Catalyst switches that allows one or more port’s IP traffic to be copiedand sent to another single destination port on the switch. The ports that are used for the input to a SPAN arereferred to as source ports. The port where all the copied traffic is sent to is called the destination port.

NOTE: (On some switches, the SPAN destination port is referred to as the monitor port. In this documentthis port will always be referred to as the destination port.) Think of SPAN as a funnel that collectsnetwork traffic from multiple ports and copies it to a single output port. The destination port of a SPANis used by Activ! IP to sniff for voice traffic to and from agent phones. Depending on the switch model,the “source ports” used by SPAN can be ports or VLANs. Only certain types of ports can be used assource ports. Using switch ports as source ports is referred to as PSPAN (Port SPAN). Using VLANs assource ports are referred to as VSPAN (VLAN SPAN). Some switches support only PSPANs. Other switchessupport both PSPANs and VSPANs. Some switches support the use of both ports and VLANS in a singleSPAN configuration. Local SPANs (LSPANs) are SPANs where all the source ports and the destinationport are physically located on the same switch. Remote SPANs (RSPANs) can include source ports thatare physically located on another attached switch. The number of SPANs that can be configured canvary by switch. SPAN configuration and functionality is not the same on all Cisco® Catalyst switches.Some switches can have the SPAN destination port configured to only show packets that are incomingto the source port(s) (ingress traffic) or only packets that are outgoing to the source port(s) (egresstraffic). The default for many switches is to show both ingress and egress packets hitting the sourceport(s). On some Catalyst switches, the destination port of a SPAN will not accept regular networktraffic. For more information on SPAN and RSPAN, refer to your switch documentation.

Software Configuration

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SPAN Tap Point Options

Port Spanning 1- Monitor traffic of all recorded phones on both transmit and receive (TX-RX). The Gateway(s)and Call Manager Server(s) do not need to be spanned if the phones are being spanned both in transmit andreceive mode. This will allow for recording of both internal extension to extension calls and external inbound andoutbound calls. Optionally, create two span ports to minimize span port traffic, as the IPX Card can accept inputfrom two span ports providing that duplicate traffic does not exist on them. Duplicate traffic would then provideduplicate events to the IPX Card which is not supported.

Port Spanning 2- Monitor the traffic of just the Call Manager Server(s) and Gateway(s) on both transmit andreceive (TX-RX). Optionally, create two span ports to minimize span traffic; one for the Gateway(s) and one for theCall Manager Server(s). This will allow for recording of only external outbound and inbound calls. Internal extensionto extension calls will contain no audio but will get call control events. Use the Ignore Internal Calls option in thisconfiguration as internal extension to extension calls will never contain audio.

VLAN Spanning- Monitor traffic of the whole voice VLAN. Normally the span can just be set for Transmit (TX)only and this is sufficient.

There are many other SPAN options depending on the exact network configuration, switches used, sites involvedetc. The configurations outlined above are just the most common.

NOTES: Monitoring the Call Manager Server and the phones is what provides the Call Control Eventsthat are required to make this solution work. Since the phones are talking directly to the Call ManagerServer audio is heard via monitoring just the phones themselves. Monitoring just the Call Manager“cable” by itself would then hear both the phones and the Call Manager and would provide eventshowever, the RTP Packets would never receive the audio. Once the call is established, the audio (RTPPackets) are transferred between the two devices involved in the call, either on another phone or onan internal extension-extension call(s), or the phone and the gateway router on inbound-outboundexternal call(s). Activ! IP must get traffic to provide both Call Control Events from the phones, and audiofrom the phones and/or gateway routers.


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IPX Setup: VPConfig > Channel Manager > IPX (Tab) > Settings (Tab):

IPX Setup:

Dynamic Mode Checkbox- Activ! IP is configurable to run in two different operating modes. This allows theconfiguration to be very flexible and to accommodate most any PBX configuration. A checkbox controls bothmodes, options are configured as follows:

Static Mode- Uncheck the “Use Dynamic Channels” checkbox to use Static Mode. In Static Mode theextension or the IP address of the phone(s) must be setup manually on the Channels Tab of VPConfig.NOTE: You cannot combine Dynamic mode with Static mode. Please reference the VPConfigGuide on how to configure Channels. This configuration is very flexible as the channels are config-ured with either the IP address and/or the Directory Number/Extension of the phone.


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When a call begins, the IP address and extension are both used to determine which Activ! IP channelshould be used to record the call. If both are configured the IP is used first, then the extension if the IP isinvalid or blank. It’s recommended to use IP whenever possible in Static Mode, especially for multi-lineconfigured phones. If a phone has two lines/extensions assigned to it, and the channels/extension columnis used to configure the Activ! IP channel, it would require an extension per channel for each multi-linephone, whereas using the IP address of the phone would only require one channel. Shared directorynumber/lines/extensions should not use extension for a channel in Static Mode, it’s not supported. Thisconfiguration needs a “unique” identifier per phone, either IP or extension, as long as it’s unique.

Dynamic Mode- Check the “Use Dynamic Channels” checkbox to use Dynamic Mode.In this mode the first unused available recording channel will be used for recording the call. This in turnallows more efficient use of the recording channels as a smaller number of recording channels are neededto record the same number of phones as Static mode. It’s limited only by the maximum number ofsimultaneous recordings which in turn is limited by the number of available channels in Activ! IP. Forexample, a 64 channel recorder can be used to record 256 phones/devices but can only record 64 callssimultaneously. When the channel capacity is exceeded by the number of simultaneous calls, recordingsare then ignored in the order received. Each call will normally use a different recording channel undernormal traffic rather than the same channel that Static Mode provides. Activ! IP by default will record everyphone that generates traffic routed to the recorder in Dynamic Mode. With this configuration channelscannot be locked down to a specific phone in any way. This mode requires no configuration for the ChannelsExtension, IP settings, or anything else for that matter to work—it’s basically plug and play in this mode.The Ignore List within VPConfig can be used in order to not record specific phones in this mode.

Log IPX Events (Trace)- This feature sends all trace events to the Event Center Log file. This includesrouting information, D-Channel Events and Call Control Events. This checkbox should only be checked fordebugging.

Save Dialed Digits- Check this option to capture digits pressed while dialing. When this option is checkedthe Digits Pressed will be saved into the ANI/NumberDialed column. The Calling Party and Numbers dialedwill be separated by a forward slash “/” when this is checked. This can be very useful in determining ifan outbound Called Party Transformation is in effect for this call. The digits pressed will show what wasactually dialed while the Called Party will show the number it was transferred to.

Ignore Internal Calls- Check this option to ignore internal extension to extension calls. This will allow foronly recording external inbound and outbound calls. Basically this provides trunkside recording functionalitybut is still stationside recording. It’s primarily used when monitoring only the Call Manager Server(s) andGateway(s) since there would be no audio present on extension-extension calls but it can also be usedwhen monitoring just the phones themselves to ignore unwanted internal calls.

# VOX Channels- Enter the number of VOX channels to be mixed with VoIP channels. All Smartworks®

analog cards can be used but must be configured as the first Board(s)/Channels in the system. Analogchannels can only be used in VOX mode, Loop Detect is not supported. Analog channels must be set to“Always Record (Vox Emulation)” within the VPConfig/Channels settings.


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Ignore Extensions/IP:

This feature is used with Dynamic Mode in order to ignore specific Extensions or IP Addresses or a mixturethereof that should not be recorded. Just highlight the extension(s) then select the Minus Sign button. Use theAdd Sign Button to re-add the extension(s).

Record Extensions/IPs:

This feature is used in Dynamic Mode to record only the devices contained in this list. All other devices calls willbe ignored.

Called and Calling Party Transformations

Cisco® Call Manager can be configured to use Called or Calling Party Transformations. Both of these featuresaffect the data passed in the Skinny Protocol as Called or Calling Party. Activ! IP can be configured to handleParty Transformations for either the Called or Calling Party. When this feature is used on the Call Manager theactual Called or Calling Party data at times is then moved to either the Called or Calling Party VM or Party Namefields within the Skinny protocol. This GUI is used to handle this feature and get the actual Called or Calling Partydata rather than the transformed data. Transformed numbers should be entered here when Acitv! IP needs to lookinto other fields for the correct data. When Activ! IP encounters any number as either Called or Calling Partywithin this list it will then automatically look into the other Skinny fields in order to locate the correct non-transformed data to use for either extension or ANI/Numberdialed. Both features and how to configure Activ! IPfor handling these features are discussed below:

Calling Party- The Calling Party is what Activ! IP uses for extension on outbound calls, and ANI oninbound calls. Normally this value would be the Directory Number/Extension of the phone that is makingthe call unless it’s being transformed by the Call Manager and/or IPCC/ICM Servers. Calling PartyTransformations will change the Calling Party to whatever the Transformation is set for therefore changingwhat Activ! IP is presented via the Skinny protocol as the Directory Number/Extension. This option allowsActiv! IP to handle Calling Party Transformations and still capture the phone’s real Directory Number/Extension to be saved in the Extension database column and pass on to other applications that may beusing Extension via the VPI API’s. Calling Party Transformations are very commonly used and it’s highlyrecommended to check for the correct data on call records to determine if the real Directory Number/Extension is being captured and Activ! IP and the Cisco Call Manager and are properly configured to getthis data. Changes within the Cisco systems involved may be needed to provide correct or desired data forthe Calling Party. If a Calling Party Transformation is used it may be necessary to put the phones DirectoryNumber/Extension into the “Display (Internal Caller ID)” section of the Line Settings of each specificDirectory Number assigned to a phone within the Call Manager Administration to provide the real DirectoryNumber/Extension. This section can accept a mixture of both numbers and letters for a value on the Call


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Manager without affecting Activ! IP as the letters are stripped from the value and just the numbers areused. For example “Stephanie-4002” could be used for the “Display (Internal Caller ID)”. Activ! IP will stripthe letters from the value and end up with “4002” as the Directory Number/Extension. It’s recommended tonot use any other numbers except the desired Directory Number to be saved as the extension when theCalling Party is transformed.

Called Party- If a Called Party Transformation is being used on the Call Manager this setting will changethe data seen via Skinny for the Called Party. The Called Party data is saved by Activ! IP into the ANI/NumberDialed column. If the Save Digits option is enabled you will see the reported Called Party along withthe number actually dialed separated by a forward slash “/”. (Ex: “8053895200/8055551212” where8055551212 is the transformed number). The GUI in VPConfig can be used to watch for transformednumbers and allows Activ! IP to look into either the CalledPartyVM or CalledPartyName columns for thevalue to use as Called Party. This setting is normally not used or needed by Activ! IP since enabling SavedDigits would show both values, the numbers dialed and the transformed number which shows as CalledParty. If Save Digits is not or cannot be used then this setting can be used to attempt to get a value otherthan Called Party if the correct value is seen in either the CalledPartyVM or CalledPartyName columns.

Called Party- The Called Party is what Activ! IP uses for extension on inbound calls, and Numberdialedon outbound calls. Normally this value would be the Directory Number/Extension of the phone that isreceiving the call unless it’s being transformed by the Call Manager Server and/or IPCC/ICM Servers.Calling Party Transformations will change the Called Party to whatever the Transformation is being set fortherefore changing what Activ! IP is presented via the Skinny protocol as the Directory Number/Extension.This option allows Activ! IP to handle Called Party Transformations and still capture the phone’s realDirectory Number/Extension to be saved in the Extension database column and pass on to other applicationsthat may be using Extension via the VPI API’s. *Called Party Transformation is rare to see on inbound callswhen monitoring stationside but can be very commonly used on outbound calls in stationside monitoring.Changes within the Cisco systems involved may be needed to provide correct or desired data for the CalledParty. If a Called Party Transformation is used it may be necessary to put the phones Directory Number/Extension into the “Display (Internal Caller ID)” section of the Line Settings of each specific DirectoryNumber assigned to a phone within the Call Manager Administration to provide the real Directory Number/Extension. If a Called Party Transformation is being used on the Call Manager this setting will change thedata seen via Skinny for the Called Party. The Called Party data is saved by Activ! IP into the ANI/NumberDialed column on outbound calls. If the Save Digits option is enabled you will see the reportedCalled Party along with the number actually dialed separated by a forward slash “/” on outbound calls.


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Party Transformation - Cisco Wildcards

Activ! IP’s Party Transformation GUI supports the use of most commonly used Cisco wildcards.The Woldcards cover a broad range of “transformed” numbers without having to type them allindividually. For example a commonly used Cisco® Call Manager transformation setting would be“805389xxxx” , this would set the Calling Party to be “8053891234” if the Directory Number of thephone initiating the call was 1234. The xxxx is replaced by the Directory Number of the phone thatdialed the call. In this case, the Calling Party Transformation in Activ! IP for example could be setfor the following values:

1. 805389xxxx would cover any number that started with 805389 and then had 4 more followingdigits. For example 8053895200 would be a match while 80538952001 would not be.

2. 805389@ would cover any number that started with 805389 with any number of followingdigits. For example 8053895200 would be a match while 80538952001 would also be a match.

Supported Cisco® Wildcards: @, X, [ ^ - ], 0 1 2 3 4 5 6 7 8 9 A B C D, *, #

@ The at symbol (@) wildcard matches all numbers. Each pattern can have only one @wildcard.Ex: The pattern 389@ matches all numbers in the range 3890000000 through3899999999.

X The X wildcard matches any single digit in the range of 0 through 9.Ex: The pattern 9XXX matches all numbers in the range 9000 through 9999.

[ ] The square bracket ([ ]) characters enclose a range of values.Ex: The pattern 813510[012345] matches all numbers in the range 8135100 through8135105.

- The hyphen (-) character, used with the square brackets, denotes a range of values.Ex: The pattern 813510[0-5] routes or blocks all numbers in the range 8135100through 8135105.

^ The circumflex (^) character used with the square brackets negates a range of values.Ensure that it is the first character following the opening bracket ([). Each pattern canhave only one ̂ character.Ex: The pattern 813510[^0-5] matches all numbers in the range 8135106 through8135109.

Unsupported Cisco® Wildcards: !, ?, +


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Calling Party Transform Ext (First and Last X Digits):

Enabled this feature to minimize either First or Last digits when using Calling Transformation for Extensions.Valid values for these fields are the numbers 1-32.

This field contains a list of discard patterns that control the discard digit instructions. For example, in asystem where users must dial 9 to make a call to the Public Switched Telephone Network (PSTN), the PreDotdiscard pattern causes the 9 to be stripped from the dialed digit string.

Called Party Transform Ext (First and Last X Digits):

Enabled this feature to minimize either First or Last digits when using Called Transformation for Extensions.Valid values for these fields are the numbers 1-32.

Calling Party Transform Masks:

This field specifies the calling party transform mask for all calls routed through this route group. Valid valuesfor this field are the numbers 0 through 9, and the wildcard character X. This field can also be left blank. If it isblank and the preceding field is set to Off, no calling party number is available for Calling Line Identification(CLID).

The calling party transform mask can contain up to 50 digits.

Add the CLID in the box provided and select the Plus (+) Sign Button. Remove a CLID using the Minus (-) SignButton.

Called Party Transform Masks:

This field specifies the called party transform mask for all calls routed through this route group. Valid values forthis field are the numbers 0 through 9, and the wildcard character X. This field can also be left blank. If thisfield is blank, no transformation takes place—the dialed digits are sent exactly as dialed.The calling party transform mask can contain up to 50 digits.

Add the CLID in the box provided and select the Plus (+) Sign Button. Remove a CLID using the Minus (-) SignButton.

Apply/Cancel Buttons:

Select these buttons prior to exiting the Settings Tab.


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IPX Setup: VPConfig > Channel Manager > IPX (Tab) > Boards (Tab):

IPX Boards:

Here is where you add additional IPX boards depending on how many channels you need to record. Use the Addand Delete Board buttons accordingly (i.e., 64 channels per IPX Card).

• Analog Cards will not appear here, this section is just for IPX boards.• The board numbering does not reflect the Smartworks® Board number.

Signaling Protocols:

Choose the appropriate terminal control protocol for your recording environment. If required, multiple signalingprotocols can be selected at the same time. For example a Cisco® PBX can use Skinny phones, SIP phones andH.323 devices. In this configuration Skinny, SIP and H.323 Signaling Protocols can be selected to record all threetypes on the same system.


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For example a Cisco® PBX can use both Skinny and SIP phones.

Cisco® Skinny- Select this protocol for use with Cisco® Call Manager or any other PBX or devices thatmay use the SCCP protocol to communicate. This protocol provides extensive information about each call,Activ! IP saves all of the information provided into custom fields which represent the actual field namesused within the Skinny protocol.

Port Settings:Protocol Type - MT_TCP or MT_UDP, the type of protocol used by the network to listen for signaling data.Default: TCPPort - Port Number reserved on the Call Manager to listen for signaling requests associated with Cisco®

telephones. Default: 2000 (Location: Cisco® Call Manager Configuration screen Setting: Ethernet PhonePort)

Supported phone models tested:Cisco® 7960Cisco® 7940Cisco® 7912Cisco® 7910Cisco® 7920 Wireless PhoneCisco® IP Communicator (PC Software Phone)NOTE: All Skinny phones and devices should be supported.

Unsupported phone models tested:Cisco® IP Softphone-No Call Control events are provided by this software phone. Recommend using theCisco® IP Communicator instead which does provide Skinny protocol Call Control events.

Software Release Version tested:Cisco® Call Manager Versions 3.3, 4.1, 4.2, 5.0

Cisco® IPCC Express- Standard, Enhanced, PremiumCisco® IPCC Enterprise 7.0Cisco® ICM 7.0 build 14833Cisco® CRS 3.5.3Cisco® Agent Desktop 7.0(0) enhanced version build 7.0.0.39If other versions are used, different behaviors may be observed that could require additionalprogramming changes.

Avaya® H.323- Select this protocol for use with Avaya® IP PBX S8300 or the Avaya® IP Office PBX.Configure the Port Type and Port Number used by the Avaya® PBX.

Port Settings:H225CS - H225 Call Signaling IP Protocol Type (TCP/UDP) and the port designated by the PBX for listeningto signaling information associated with IP phones. By default, the TCP protocol is used on port 1720.

H225RAS - H225 Registration Admission and Status IP Protocol Type and Port (if RAS is not enabled onthe network than these fields must be ‘NULL’). NOTE: RAS must be enabled and configured correctlyin order to obtain caller and called phone numbers via the IPX.


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Phone models tested:Avaya S83004620461246064602IP Office56024620SW5620SW4610SW4602

Software Release Version tested:Avaya® IP PBX S8300 G3xV11, Comm. Manager 1.3Avaya® IP Office PBX - 3.0If other versions are used, different behaviors may be observed.

H.323- Select this protocol for recording standard H.323 protocols. This is commonly used for the Cisco®

Call Manager to communicate to a Gateway router and would be used if the Tap Point was the cable goingto the Gateway.

Nortel® UNIStim- Select this protocol to record the Nortel® Meridian 1 PBX or the Nortel® BusinessCommunications Manager (BCM).Port Settings:LTPS Port Number - the number of the port used by the Nortel® IP PBX for listening to signaling requestsfrom the VoIP endpoints. By default, the Nortel® Meridian 1 uses port 5100 while the Nortel® BCM uses port7000.

LTPS Type - MT_TCP or MT_UDP, the type of protocol used by the network. By default, the Nortel® PBXrelies on the UDP protocol.

IP Phone Port - Port Number reserved on the IP phone to listen to signaling requests from the PBX. Bydefault, Nortel® phones reserve port 5000.IP Phone Type - MT_TCP or MT_UDP, the type of protocol used by the network. By default, the Nortel®PBX relies on the UDP protocol.Phone models tested:

Meridian 1Series ModelIP2004 NTDU92 TIP2002 NTDU91 TIP2007 NTDU96 TIP 1120E NTYS03 TIP2001 NTDU90 T


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BCMSeries ModelIP004 NTDU82 T

Software Release Version tested:Nortel® Meridian 1 with Option 11c release version 4.Nortel® Business Communications Manager (BCM) Release 3.6 Build 2.2c.If other versions are used, different behaviors may be observed.

Ericson® H.323- Select this protocol to record the Ericsson® MD110 PBX.

Port Settings:H225CS - H225 Call Signaling IP Protocol Type (TCP/UDP) and the port designated by the PBX for listeningto signaling information associated with IP phones. By default, the TCP protocol used is on port 1720.

H225RAS - H225 Registration Admission and Status IP Protocol Type and Port. These fields are requiredwhen decoding an Ericsson® PBX. By default the UDP protocol is used on port number 1719.

Phone models tested:DIALOG® 4422 TDIALOG® 4425 T

Software Release Version tested:Ericsson® MD110 versionBC 12 build CXP1010130/2/BC12SP6/R3BIf other versions are used, different behaviors may be observed.

SIP- Select this protocol to record standard SIP phones or devices.Phone models tested:X-Lite softphone® - v2.0 rel. 1103mAvaya One -X®, and all phone models that support this firmware*Asterisk Grandstream®: Budget tone 100, GXP 2000Asterisk Snom® 320Asterisk Zulty® Zip2Asterisk Linksys® SP841

* The IPX is capable of decoding the SIP protocol on networks using this PBX and phone models,however, the media (RTP) data is either encrypted or scrambled.Siemens®- Select this protocol to record the Siemens HiPath 4000 PBX.

Port Settings:Two Transports (IPPROTO and Port fields) are needed to enable Siemens with the IPX platform. Thecontrol signaling is proprietary and we have seen it running on port 4060 over TCP. The media is establishedusing h323 on an h225\h245 channel which we have seen running on port 1720 of TCP.


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Phone models tested:Optipoint 410 advance,Optipoint 410 economy plus,Optipoint 420 advance,Optipoint 410 entry

Software Release Version tested:HiPath 4000 Ver 3.0 SMR5 SMP4If other versions are used, different behaviors may be observed.

Alcatel®- Select this protocol to record the Alcatel® OMNI PCX Enterprise Systems.

Port Settings:One Transport is needed to enable Alcatel with the IPX platform. All of the signaling is proprietary and wehave seen it running on port 32640 over UDP.

Phone models tested:4038 IP Touch Set US (8 Series)4068 IP Touch Set US (8 Series)4035 IP Advanced e-reflexes4020 IP Premium e-reflexes4010 IP Easy e-reflexesSoftware Release Version tested:OMNI PCX Enterprise 6.0If other versions are used, different behaviors may be observed.

Board Protocol (i.e., Cisco®):

Depending on the Signaling Protocol, select either TCP (i.e., Cisco Skinny) or UDP (User Datagram Protocol, notnormally used but it is supported).


Select these buttons prior to exiting the Boards Tab.


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VPConfig > Channel Manager > Database (Tab):

Activ! IP uses several new database columns for VoIP that are not contained within the default database. Thissection provides details on these new database columns.

Database Mappings:

The GUI shows the database columns that can be used when the VoIP Channel Manager is enabled and alsoallows for the creation of new database columns. If need necessary, the mapping of these columns can bechanged. The left most entries show the columns that Activ! IP will use to save the data, the right most entriesshow the database column that will be used to hold the data. If no database column exist there will be nomapping shown. In this case use the Database Field + button to create a new column for the data.


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Database Field:

This allows for the creation of and modification of the new database columns mappings. Select an entry from theDatabase Mappings menu, then click the + button to the right to create the new column. Optionally, the mappingfor any entry in the Database Mappings can be changed here. Normally there is no need to modify the DatabaseMappings unless some custom configuration is required.


Select these buttons prior to exiting the DatabaseTab.


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VPConfig > Channel Manager > Software RTP (Tab):

Activ! IP uses a standard network as the RTP recording device. VPConfig will show this Configuration Tabwithin the Channel Manager called Software RTP. This tab contains all of the configuration options for theSoftware RTP capture within Activ! IP.

NOTE: Enable License Activ! IP for D-channel DLL and set registry setting (DSP Hardware=6)

RTP Setup:

Destination IP Address- This configures the IP Address that Activ! IP will both listen on and stream the RTPdata to and from the IPX card. It must be set for the same subnet as the IPX cards output (port 2) or no audiowill be recorded.

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Time Intervals:

Idle Timeout: (integer) default 5 – Time value in seconds after each a channel is declared idle and stopsrecording

InterPacket Sleep: (integer) default 100 – Time value in milliseconds between polling RTP receive ports. Designedto improve performance in the future.

Ports:

Left Port: (integer) default 4000 – Left channel UDP port to receive RTP left channel traffic.

Right Port: (integer) default 6000 – Left channel UDP port to receive RTP right channel traffic.

RTP Codec:

Select the type of coding/decoding to be performed on the data stream, here are some examples:

• G.711• G.729a


Select these buttons prior to exiting the Software RTP Tab.


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Skinny Protocol Database Fields

Skinny Protocol Database Fields

This section outlines the Skinny Protocol’s internal communication fields and their definitions. This data providesdetailed information about the call all of which is captured and saved to Activ! IP’s database.

CallingPartyName- When a name can be associated with this number (CallerID when possible, or configuredvia PBX).

CallingParty- The number of the caller.

CalledPartyName- When a name can be associated with this number (CallerID when possible, or configured viaPBX).

CalledParty-The number that was originally dialed. LineInstance-which phone line the call is active on. This isphone model dependant.

CallId- A unique number assigned by the IPX to this call. This number is unique per each Station and is notunique to the complete system. Same as call ref number provided with the call control events reported by the IPX.

CallType- The direction and type of the call. Values below:INBOUND=1OUTBOUND=2FORWARD=3

OrigCalledPartyNumber- The number that was originally dialed. For example, this field is relevant when theoriginal number is configured for call forwarding.

OrigCalledParty- This is relevant when call has been redirected or forwarded.

LastRedirectingPartyName- When the call has been re-directed, this is the name associated with the re-directing phone number (when available).

LastRedirectingParty- When the call has been re-directed, this is the phone number which re-directed the call.

OrigCalledPartyRedirectingReason- The reason the original called number was redirected on the previous callsegment. This would indicate this call has another part associated with it.

LastRedirectReason- The reason the last call was re-directed.

CallingPartyVoiceMailbox- When directed to Voicemail, the mailbox number used by the calling party.

CalledPartyVoiceMailbox- When directed to Voicemail, the mailbox number used by the called party.

OriginalCalledPartyVoiceMailbox- When the call has been forwarded, the original VoiceMailbox used by thecalled party.

LastRedirectVoiceMailbox- When the call has been re-directed, the original VoiceMailbox used by the calledparty

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Cisco® Skinny Redirect Reasons

The Redirect Reasons provided within the Skinny Protocol provides more detailed and useful information about eachtransferred, diverted, or forwarded call. There are two fields provided in Skinny that use these Redirect Reasons, theyare called OrigCalledRedirReason and LastRedirReason. If a value is present within the OrigCalledRedirReason fieldthis would indicate that there is a previous call somehow associated with this call and the value shown wouldindicate why and how it was redirected. See chart below showing all Skinny Redirect Reasons and theirdescriptions.

The OrigCalledParty, OrigCalledPartyName and OrigCalledPartyVM fields would provide the CalledPartyDatadata for this preceding “part” of the call. A value in LastRedirReason would indicate why and how this call itselfwas redirected and the LastRedirParty, LastRedirPartyName and LastRedirParty VM fields would contain thedata showing where this call was redirected from. Using this data all “parts” of a call can be located. If no data isprovided for either RedirReason there is no other call(s) associated with this call. Unfortunately not enough datais provided for Activ! IP to perform the FindAllParts feature to locate any other previous parts of a given call. Thereis more than enough data to do this manually and locate all recorded parts.

Value Description

Q.931 Standard Redirect Reason Codes

0 Unknown

1 Call Forward Busy

2 Call Forward No Answer

3 Announced Transferred Call

4 Call Transfer

5 Call Pickup

7 Call Park

8 Call Park Pickup

9 CPE Out of Order

10 Call Forward

11 Call Park Reversion

15 Call Forward All

Non Standard Redirect Reason Codes

256 Call Deflection

512 Blind Transfer

768 Call Immediate Divert

1024 Call Forward Alternate Party

1280 Call Forward On Failure

1536 Conference

1792 Barge

Cisco® Skinny Redirect Reasons

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Cisco® Specific Features Supported

Barge- Allows you to add yourself to non-private calls on a shared phone line. Barge features include cBarge andBarge. Typically, your System Administrator will provide only one of these barge features to you:

• cBarge adds you to a call and converts it into a conference, allowing you and other parties toaccess conference features.

• Barge adds you to a call but does not convert the call into a conference. Your system administratormust configure Barge features for you. Associated softkeys: cBarge, Barge

NOTES: Activ! IP has been tested to work with Cisco’s Barge and cBarge features. Barge results in asecond recording for the extension performing the barge.

Call Forward- Allows you to redirect your incoming calls to another number.Associated softkey: CFwdALLAssociated key sequence: **1

NOTES: Activ! IP has been tested to work with forwarded calls. Call records will show the extension ofthe phone that answered the call as the Number Dialed. The original called party is kept in the CalledParty column.

Call Park- Allows you to park (temporarily store) a call and then retrieve the call by using another phone in theCisco Call Manager system. Call Park can be useful if you want to transfer a call from your phone to a phone ina lab or conference room, for example: Associated softkey: Park

NOTES: Activ! IP has been tested to work with Call Park. When a call is parked, the Call Managertreats this just like a hold at first. Recording is stopped if the call is parked.

Call Pickup- Allows you to redirect a call that is ringing on another phone to your own phone, so you can answerthe call. Call Pickup features include Pickup, GPickup, and OPickup:

• Pickup allows you to answer a call that is ringing on another phone within your “group” (a collectionof extensions that your System Administrator defines).

• GPickup allows you to answer a call ringing on a phone in another group.

• OPickup allows you to answer a call ringing on a phone in another group that is associated withyour group. Call Pickup can be useful if you share call-handling tasks with coworkers.Associated softkeys: PickUp, GPickUp, OPickUp

NOTES: Activ! IP has been tested to work with Call Pickup. When a call is picked up from Park therewill be a call record showing an outgoing call from the extension to the pilot number associated withthe Parked Call.

Caller ID- Displays caller-identification, such as a phone number, name, or other descriptive text, on your phonescreen.


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NOTES: Activ! IP has been tested to work with Callerid. Callerid is captured and saved if provided to thephone.

Call Waiting- Indicates (and allows you to answer) an incoming call that you receive while on another call. Callwaiting also displays incoming call information on your phone screen.Associated softkey: Answer

NOTES: Activ! IP has been tested to work with Call Waiting. When answering another call the existingcall is placed on hold by the Call Manager which causes recording to stop. When the next call isanswered a new recording will then begin for the Call Waiting call.

Conference Features- Allows you to talk simultaneously with multiple parties.Conference features include Conference, Join, cBarge, and Meet-Me:

• Conference (or ad-hoc conference) allows you to initiate a conference by calling each participant.• Join allows you to connect current callers who are on a single line by creating a conference call.• cBarge allows you to establish a conference by adding yourself to a call on a shared phone line.• Meet-Me allows you to call a predetermined number at a scheduled time to host or join a conference.

Associated softkeys: Confrn, Conference, Join, cBarge, MeetMe

NOTES: Activ! IP has been tested to work with all types of conferences mentioned here. Any pilotnumbers and/or conference I.D.’s are captured as part of the call(s) record(s). Normally a b00 will bepreceded by the internal conference bridge associated with a conference and normally will bedisplayed as Called Party. This information can be used to manually determine all parties recordedwithin a conference call.

Direct Transfer and Announced Transfers- Allows you to connect two calls to each other (without remainingon the line yourself).Associated softkey: DirTrfr

NOTES: Activ! IP has been tested to work with Direct Transfers. All calls involved in a direct transferwill be recorded. The actual transferred call may not be marked as a transfer due to the fact that onsome versions of Cisco Call Manager there is no reliable way to determine when and if the transferactually took place. Announced Transfers have a LastRedirectReason that provides a value to determineif a call was transferred but Direct does not. Activ! IP marks calls as transferred if the LastRedirectReasoncolumn contains a value that is seen as a transferred call. See Cisco Skinny Redirect Reasons for moreinformation on translating Redirect Reasons.

Extension Mobility Service- Allows you temporarily to apply your phone number and user profile settings to ashared Cisco IP Phone by logging into the Extension Mobility service on that phone. Extension Mobility can beuseful if you work from a variety of locations within your company or if you share a workspace with coworkers.Associated button: Services


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NOTES: Activ! IP has been tested to work with Extension Mobility. Since Activ! IP captures the callinformation on a per call basis, the directory number/extension change that “will follow” the useridlogged into a given phone, will be captured correctly for all calls made after the login. Unfortunatelythe userid itself cannot be captured by monitoring the Skinny Protocol data and does not get capturedby Activ! IP. The login itself uses HTTP protocol on port 80 in order to login to the Call Manager. Noother data provided is utilized to determine the userid that logged into a phone. Use the userid’sassociated-assigned Directory Number to “follow” an agent in a free seating environment.

Hold- Allows you to move a connected call from an active state to a held state.Note that your phone allows only one call to be active at a time; other calls will be put on Hold.Associated button: HoldAssociated softkey: Hold

NOTES: Activ! IP has been tested to work with Held Calls. Calls put on Hold will automatically stoprecording. This is not configurable. A MediaStopped event is given with the Call Held event.MediaStopped cannot be ignored, therefore calls put on hold will always stop recording.

Immediate Divert- Allows you to transfer a ringing, connected, or held call directly to your voice-messagingsystem.Associated softkey: iDivert

NOTES: Activ! IP has been tested to work with Immediate Divert. The device/phone taking the call willbe saved as the Ani/Numberdialed. The OriginalCallPartynumber and Name columns will hold theactual directory Number/Extension that was dialed before it was Diverted. Diverted calls may get markedas transferred if the LastRedirectReason=4.

Join- Allows you to join two or more calls that are on one line to create a conference call. You remain on the call.Associated softkey: Join

NOTES: Activ! IP has been tested to work with Join. If a recorded phone Joins a call, another recordingwill be started for the Joined phone.

Meet-Me Conference- Allows you to host a Meet-Me conference in which other participants call a predeterminednumber at a scheduled time. To host a Meet-Me conference, press MeetMe and dial the Meet-Me conferencenumber (provided by your system administrator). Meet-Me participants do not use the softkey but simply dial thenumber at the scheduled time. Participants get a busy signal if they call the conference number before the hosthas dialed in.Associated softkey: MeetMe

NOTES: Activ! IP has been tested to work with Meet-Me-Conferences.

Multiple Calls per Line Appearance- Allows each line on your phone to support multiple calls. The defaultconfiguration specifies four calls per phone line, but your system administrator can adjust this setting. Only onecall can be active at any time; other calls will be on hold.

NOTES: Activ! IP has been tested to work with Multiple Calls Per Line Appearance. When moving fromcall to call each “active” call will become a new recording with it’s own call data. Switching from callto call is really treated like a call put on hold. Recording stops, and a new call begins.


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Multiple Lines per Phone- Allows you to handle calls on multiple phone lines. Your system administratorassigns one or more phone lines (directory numbers) to your phone.

NOTES: Activ! IP has been tested to work with Multiple Lines Per Phone. When moving from line toline each “active” call on each line will become a new recording with it’s own call data. Switchingfrom line to line is really treated like a call put on hold. The recording stops, and a new call begins.

Phone Line Text Label- Allows you to create a text label that shows up on your phone screen for each of yourextensions or phone lines. This feature can be useful if you have multiple lines on your phone. The line labelsetting can be accessed from the User Options web pages.

NOTES: Activ! IP has been tested to work with Phone Line Text Labels. This feature normally addsdata put here to the Called/Calling Party Name field within the Skinny Protocol and is captured byActiv! IP.

Programmable Buttons- Depending on configuration, programmable buttons provide access to:

• Phone lines (line buttons)• Speed-dial numbers (speed-dial buttons)• Web-based phone services (for example, a Personal Address Book button or Fast Dials button)• Phone features (for example, a Privacy button)

Your system administrator determines how many lines (and therefore line buttons) you have on your phone; he orshe can configure other types of programmable buttons for your phone as well.Using your User Options web pages, you can assign speed-dial numbers or phone services to availableprogrammable buttons. Your system administrator might need to first configure a Service URL button for yourphone.

NOTES: Activ! IP has been tested to work with Programmable Buttons. None of these features appearto have any affect on the Skinny Protocol data therefore have no affect on the operation of Activ! IP.

Redial- Allows you to call the most recently dialed phone number by pressing a button.Associated softkey: Redial

NOTES: Activ! IP has been tested to work with Redial. This feature has no affect on getting thenumberdialed/called party information. The only difference are digits not being pressed and a softkeybutton pressed, neither of which have any affect at all to the operation or data capture by Activ! IP.

Remove Conference Participants- Allows the conference initiator to drop participants from the conference callby using Remove or Remove Last Conference Participant:

• Remove drops the selected participant• Remove Last Conference Participant drops the most recently added participant

Associated softkeys: Remove, RmLstC

NOTES: Activ! IP has been tested to work with Remove Conference Participants. If a recorded phone isremoved from a conference, recording is stopped just for the phone that that was removed.


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Resume- Allows you to resume a call that you had put on hold.Associated button: HoldAssociated softkey: Resume

NOTES: Activ! IP has been tested to work with the Resume Call feature. If a recorded phone resumesa call a new recording will start.Shared Line- Allows you to have multiple phones that share the same phone number or allows you to share aphone number with a coworker. Shared lines can use special features such as Barge and Privacy.

NOTES: Activ! IP has been tested to work with Shared Lines. Activ! IP does not track calls or useextensions internally to record calls. Calls are tracked by sessionid so phones with the same DirectoryNumber/Extension do not affect the operation of Activ! IP. However, using Static Mode and having anExtension only per channel will be affected. To use Static Mode with Shared Lines it’s recommended touse the IP address per channel rather than the Directory Number/Extension.

Speaker Mode (listen-only)- Allows you to listen to a phone call hands-free (without using the handset).Associated softkey: Monitor Speakerphone ModeAssociated button: Speaker

NOTES: Activ! IP has been tested to work with Speakerphone Modes. Activ! IP records the audiodirectly from the transmission between the two phones/devices and is not concerned which deviceon the phone is used. Handset, Speakerphones, Headsets and any other supported communicationdevice is supported on the Cisco phones.

Speed Dialing- Allows you to enter an index code, press a button, or select a phone screen item to place a call(rather than dialing the number manually). You can use your User Options web pages to assign a speed-dialnumber to a programmable phone button or to an Abbreviated Dialing index code (1-99).Associated buttons: speed-dial button, assigned keypad button

NOTES: Activ! IP has been tested to work with Speed Dialing. Speed Dialing has no affect on the Called/Calling Party information and therefore has no affect on the operation of Activ! IP.

Transfer- Allows you to redirect a connected call from your phone to a another number. Transfer features includeTransfer and Direct Transfer:

• Transfer allows you to redirect a single call to a new number, with or without consulting the transferrecipient.

• Direct Transfer allows you to transfer two calls to each other (without remaining on the lineyourself).

Associated softkeys: Transfer, Trnsfer, Transf, DirTrfr

NOTES: Activ! IP has been tested to work with both types of Transferred Calls. If a recorded phone picksup a Transferred Call a new recording is started. When a call is transferred from a recorded phone therecording is stopped.


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Activ! IP Known Technical Limitations

This section outlines any known technical limitations within Activ! IP or it’s supported PBX Systems.

CRITICAL DISCLAIMER!

Cisco® Call Manager Systems- VPI cannot guarantee that data obtained on a per call basis will be 100%accurate due to the overall flexibility of the Cisco® Call Manager Systems unless it’s properly installedand configured. There are many known custom configuration options within the Call Manager that canaffect the locations and/or the data itself in the Skinny Protocol. All known custom configurations that doaffect the data provided via the Skinny Protocol used by the Cisco® Call Manager are discussed below:

Directory Number/Extension Limitation- Activ! IP can only handle directory numbers/extensions up to 9characters long due to a database limitation of the extensionnum column and the internal handling of the extensionsas an integer within Activ! IP. If a directory number/extension longer than 9 characters is encountered it isassumed that this is a transformed number and Activ! IP then looks to the Called or Calling Party VM or PartyName fields in order to get the data to use as an extension. If the extension is “transformed” the column“ExtensionTransformed” will be set to 1 on the calls record. If no usable data exists in either of these fields Activ!IP will then strip characters from the beginning of the reported data until it is 9 characters long and use this as theextension on the call record. Basically this prohibits Activ! IP from supporting more than 9 digits for any PBX asthe extension or directory number. The Cisco® Call Manager can also allow for non-numeric characters to be partof the Directory Number/Extension, for example the # and * are both valid characters on the Cisco® Call Manager.These characters cannot be saved as the extension number because they are non-numeric (not integer characters).They are automatically removed in order to be saved in the extension column of the database. The Called orCalling Party columns will contain the “un-parsed” original data provided by the Call Manager for each call record.Activ! IP’s operation of recording the audio is not affected by what the extensions are reported as. Even phonesthat have shared line/extensions/instances are supported. Calls made on shared lines/extensions do appear tothe recorder as the same extension. This can cause some strange results for example if using Static Mode andshared lines extension is used for a channel. It’s never recommended to use a shared directory number/extensionas a unique identifier.

Internal Calls- Internal calls between two recorded phones will only result in one recording for the phone thatinitiated the call. The call will show outbound from the extension that made the call to the phone that took the call.This is due to the fact that there is only one audio “path” for internal calls.

If a non-recorded phone calls a recorded phone the recording will show an inbound call with the extension thattook the call from the extension that made the call.

Cisco® and CallRefid’s- Each call on the Cisco Call Manager has two separate CallRefid’s associated witheach call, one for each side or party within the call. They are named Destination Call Leg Identifier and OriginatingCall Leg Identifier and are the same I.D.’s that are saved to the CDR database when CDR is enabled on the CallManager. *(This IS NOT the Global Callid used within the Call Manager for each call. Cisco’s Global Callid is notprovided within the Skinny data stream between the phones and PBX) External calls (incoming or outgoing callsfrom outside involving the Gateway/Trunkline Router) only provide one of these CallRefid’s via the Skinny protocol.The CallRefid for the Gateway “side” of the call is just not provided in the data stream anywhere. Both of theseCallRefid’s are saved by Activ! IP, one to CTI_CALLID and the other to CTI_NEWCALLID.Activ! IP has a feature called FindAllParts that was designed to locate all of the parts of a transferred call viausing any provided CallRefid’s from the PBX. This feature only partially works on the Cisco® Call Manager due tothe fact the CallRefid is not provided on “external” calls. FindAllParts will only find the part(s) that are directlyassociated with the actual transfer, the first incoming “external” part of the call will not be found using this feature.


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It can still be useful in finding the initiating call of an announced transfer.Cisco® Transformed Numbers- The Cisco® Call Manager has a feature called Party Transformation that canaffect what is provided via the Skinny data stream for Called and Calling Party. This can be configured on a perdevice basis or globally via a Route Pattern on the Cisco® Call Manager. This feature is commonly used in orderto mask or change the callerid information provided on either outbound or inbound calls both internal and external.This causes Activ! IP to see these transformed numbers for the directory number/extension or the ANI. Whenthis feature is used the real un-transformed data is moved either to the Called or Calling Party VM or Party Namecolumns within the data stream. NOTE: See the Called/Calling Party Transformation section for additionalinfo.

SPAN Port Limitations- Most Cisco® switches cannot handle spanning the whole VLAN or all of the ports on theswitch correctly when the utilization exceeds 50% on the switch. Packets can be lost when exceeding therecommended traffic out of a Span Port. This should always be verified for the model switch and the intendedSPAN Port configuration. It’s always recommended to create the SPAN Port(s) in order to provide the least ofamount of traffic to the SPAN port as necessary. Minimizing traffic to the SPAN port and to the Activ! IP systemwill always make for a more efficient reliable configuration.

IPX Card Limitations- With version 3.8 of Smartworks® the IPX card supports 64 simultaneous calls per card.It’s capable of monitoring 256 phones/devices. These capabilities will be increased in future Smartworks® releases.


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Index

A

Alcatel® 25Avaya H.323 22

B

Barge- 32

C

Call Forward 32Call Park 32Call Pickup 32Call Waiting- 33Called Party 18CalledParty 30CalledPartyName 30CalledPartyVoiceMailbox 30Caller ID- 32CallId 30CallingParty 30CallingPartyName 30CallingPartyVoiceMailbox 30CallType 30Cisco® IP Softphone 22Cisco® and CallRefid’s 37Cisco® Catalyst switches 13Cisco® Skinny 22Conference Features 33CRITICAL DISCLAIMER! 37

D

Destination Port 13Direct Transfer and Announced Transfers 33Directory Number/Extension Limitation 37Dynamic Mode 16Dynamic Mode Checkbox- 15

E

Ericson® H.323 24Extension Mobility Service 33

F

FindAllParts 31

H

H.323 23

I

Idle Timeout 29Ignore Internal Calls 16Immediate Divert 34Internal Calls 37InterPacket Sleep 29IPX Card Limitations 38

J

Join 34

L

LastRedirectingParty 30LastRedirectingPartyName 30LastRedirectReason 30LastRedirectVoiceMailbox 30LastRedirParty 31LastRedirPartyName 31LastRedirReason 31Left Port 29Log IPX Events (Trace) 16Loop Detect 16LSPANs 13

M

Meet-Me Conference 34Multiple Calls per Line Appearance 34

N

Nortel® UNIStim 23

O

OrigCalledParty 30, 31OrigCalledPartyName 31OrigCalledPartyNumber 30OrigCalledPartyRedirectingReason 30OrigCalledPartyVM 31OrigCalledRedirReason 31OriginalCalledPartyVoiceMailbox 30

P

Phone Line Text Label 35Port Spanning 1 14

Index

40


Port Spanning 2 14Programmable Buttons 35PSPAN 13PSPANs 13

R

Redial 35RedirReason 31Remove Conference Participants 35Resume 36Right Port 29RSPANs 13

S

Save Dialed Digits 16Shared Line 36Siemens® 24SIP 24Source Ports 13SPAN Port Limitations 38Speaker Mode (listen-only) 36Speed Dialing 36Static Mode 15Supported Cisco wildcards 19

T

Transfer 36

U

Unsupported Cisco Wildcards 19

V

VLAN Spanning 14VLANs 13VOX Channels 16VSPAN 13

Index

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Revision History

Rev. History

Page 17 added more detail to the IPX board settings. Page 21 added more detail to Called andCalling Party Transformations.

Rev A:

Pages 13 through 36 completely rewritten. Removed all references to IPM-260 Card (nowusing Standard Network Cards). Detailed information included for all Signaling Protocols,including current phone models tested/supported.

Rev Level: Date: Description:

11/20/06

Rev B:

Rev C:

03/07/07

03/14/07

Initial Release.

Rev D: 03/30/07 Added new information on new features for the IPX Board—including but not limited toCalling/Caller Party Transform Masking. Software RTP Tab added RTP Codecs. Revised IPKnown Technical Limitations.

Documents

ACTIV! IP - CHANNEL MANAGER CONFIGURATION …_IP_Channel_Manager.pdf · 4 Activ! IP - Channel Manager (rev D) Introduction Activ! IP’s recording solutions are compatible and integrate