As of February 1, 2012, Genesys is no longer an affiliate of Alcatel-Lucent; any indication of such affiliation within Genesys products or packaging is no longer applicable. Please see the Genesys website at http://www.genesyslab.com for more details.
This release note applies to all 8.1 releases of Genesys Voice Platform (GVP)
Call Control Platform (CCP).
Genesys follows applicable third-party redistribution policies to the extent that Genesys solutions utilize third-party functionality. For additional information about third-party software used in this product, see the Read Me. Contact your customer care representative if you have any questions.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. This section describes new features that were introduced in this release of Call Control Platform.
No corrections or modifications were made in this release.
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. This section describes new features that were introduced in this release of Call Control Platform.
This release includes the following corrections and modifications:
CCP no longer terminates when it fails to send the REFER
request
during transfer. (ER# 313776382)
Certain offerless non-PRACK INVITE
scenarios no longer cause CCP
to terminate. (ER# 307029642, 307029658)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. This section describes new features that were introduced in this release of Call Control Platform.
This release includes the following corrections and modifications:
The Call Control Platform does not terminate when REFER fails to send. (ER# 312184814)
An error on the Reporting Server no longer causes a connected Media Control Platform, Call Control Platform and Resource Manager to terminate under certain conditions. (ER# 289113621)
The Voice Platform (VP) Call Control Platform can now correctly handle NOTIFY
sipfrag
messages. (ER# 287469796)
When the VP Call Control Platform (CCP) is started, the build information is printed in its log file in the following order:
Note that in the Linux operating system, if the ident
utility
is not installed, then the CCP versions are not printed. (ER# 284220141)
The VP Call Control Platform no longer terminates when the CCXML application
calls destroyconference
before disconnecting all conference participants.
(ER# 283142497)
The VP Call Control Platform no longer terminates after failing to send an
outbound INVITE
. (ER# 283616012)
When the VP Call Control Platform logs messages to the Windows Event Viewer,
this unnecessary warning message no longer appears: The description for
Event ID ( X ) in Source ( XXX ) cannot be found
. (ER# 282944596)
The VP Call Control Platform (CCP) no longer creates a Windows registry entry each time it is started. Now, one entry is created the first time that CCP starts after it has been installed. (ER# 281806924)
The VP Call Control Platform no longer incorrectly sets the port to zero in
the outgoing INVITE
's Via
header when sip.transport.x
is configured with an actual IP address. (ER# 281245911)
The Call Control Platform's (CCP's) basicHTTP
event IO processor
is now case-insensitive. Previously, when header names did not exactly match
RFC 2616, the CCP ignored the posted event. (ER# 280311674)
The Media Control Platform (MCP), Call Control Platform (CCP), and Resource Manager (RM) will each write data locally to its disk when it detects some unusual delays while sending the data to the Reporting Server. Previously, the MCP, CCP and the RS would only write data locally to its disk when the amount of unsent data exceeded a certain threshold. (ER# 282743311)
The Media Control Platform's Fetching Module (FM) can now fetch a document correctly after the CURL library reconnects to the HTTP server and sends back the reconnection information to the FM. Previously, when a relative URL was used, this situation resulted in a fetching error, due to omission of the parent portion of the URL. (ER# 274249686)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following correction and modification:
The Call Control Platform can now terminate a CCXML session normally when a new page is compiled at the same time. Previously in this scenario, the Call Control Platform terminated unexpectedly. (ER# 293913307)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. There are no new features in this release of Voice Platform Call Control Platform (CCP).
This release includes the following correction and modification:
The Call Control Platform no longer terminates unexpectedly when it receives SIP RE-INVITE
requests with Session Description Protocols (SDP) that contain fewer m=
lines than the previously negotiated SDP. (ER# 277047096)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. This section describes new features that were introduced in this release of Voice Platform Call Control Platform (CCP).
This release includes the following corrections and modifications:
The Call Control Platform now correctly fetches the page when a delay is used in a send
tag and the targettype
is basic HTTP. Previously in this scenario, the platform would remain idle waiting for an event, rather than fetch the page. (ER# 266951211)
The Call Control Platform no longer sends a connection.merged
event and a connection.disconnected
event to the application when the <disconnect>
event occurs immediately after the
<merge>
event within the same transition. Previously in this scenario, the CCP did not send two connection.merged
events for each connection. (ER# 183951622)
The Call Control Platform now sends the conference.unjoined
event for the initial bridge (A<->B) if an initial bridge (A<->B) is overridden by a CCXML join (join A<-B),
and one of the endpoints (B) disconnects in the middle of receiving a re-INVITE
to establish the new bridge. (ER# 174561941)
The logging path for the ccilib.*.log
has been changed in the log.all
configuration option so that logs are now stored in a non-GVP directory. (ER# 268953628)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. This section describes new features that were introduced in this release of Voice Platform Call Control Platform (CCP).
This release includes the following corrections and modifications:
The Genesys Voice Platform 8.1 User's Guide has been updated with the following information:
The task summary table, Task Summary: Configuring the Call Control Platform, now includes:
The W3C CCXML Specification, Draft 29, specifies that the hints attribute in <createcall> and <createconference> be an ECMAScript object. There is no such specification for <dialogprepare>, however, Genesys recommends that hints are not passed as any other primitive data type for any of these three hints attributes.
(ER# 244764357)
The Call Control Platform now correctly parses device profile properties when they are specified without the comments field. Previously in this scenario, the Call Control Platform obtained an incorrect value for the property. (ER# 224262631)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
The Call Control Platform now correctly handles those CCXML sessions in which
the number of CCXML dialogs exceed the number of VoiceXML applications per session
that is configured in the IVR Profile (in the voicexml-usage-limit-per-session
parameter). Previously in this scenario, the Call Control Platform terminated
unexpectedly.(ER# 256910992)
The Call Control Platform now successfully processes multiple <dialogterminate>
requests on the same dialog. Previously in this scenario, when the dialog was
terminated the second time, it caused the Call Control Platform to terminate
unexpectedly. (ER# 252114495)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following correction and modification:
The Call Control Platform now deletes the conference objects correctly under race conditions when the conference leg is terminated from the user end and from the CCXML application simultaneously. Previously, a conference object leak and memory leak could occur under these conditions. (ER# 252851985)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following corrections and modifications:
The Call Control Platform (CCP) now correctly handles the situation where a connecting call or a conference call is terminated from user end and from the CCXML application simultaneously. Previously in these scenario, the CCP would sometimes terminate unexpectedly. (ER# 252114400, 252295300, 252683445, 253221751, 253222130, 253559694)
CCP now cleans the outstanding HTTP fetching objects before it ends the application. Previously, CCP may have incorrectly handled the outstanding requests which caused memory growth. (ER# 251887445)
CCP now correctly handles the situation when an INFO
request times out. Previously in this scenario, instead of generating an error, the CCP continued the transition to a subsequent operation (for example, <join>
), and terminated unexpectedly. (ER# 253122221)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. There are no new features in this release of Voice Platform Call Control Platform (CCP).
This release includes the following corrections and modifications:
The fetching module of the Call Control Platform (CCP) now fetches URLs that do not end with a slash (/
). Previously, when the fetching module attempted to fetch a URL in the http://www.<address>.<com>
format, the CCP terminated unexpectedly. (ER# 247567343)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. This section describes new features that were introduced in this release of Voice Platform Call Control Platform (CCP).
This release includes the following corrections and modifications:
The internal conference memory object no longer leaks under load if <conferencedestroy> is issued at the same time <disconnect> or <exit/> is issued on the connection/dialog that is currently joined to the conference. Previously, the CCXML application had to be written so that the <conferencedestroy> was issued last, after all of the participants were unjoined from the conference. (ER# 230975146)
An incoming call no longer fails to join to other end points when the Audio Codes Mediant Gateway has either SRTP or FAX enabled, and the call lands on either CCP or CTI Connector. Previously, the Mediant Gateway had to be configured to disable both SRTP and FAX in the incoming call. (ER# 227395971, 227446083)
When the remote end (UAS) sends the offer in 200 OK, the Call Control Platform is forced to provide an answer in ACK. If the offer is invalid and the corresponding answer is not generated, the Call Control Platform now sends an ACK and waits for the SIP transaction timer to clean up the call. Previously, to workaround this issue, you had to set the offer-less-invite device profile parameter to false for this device, so that the offer/answer occurred between INVITE and 200 OK. (ER# 226950236)
The reason phrase is now present if the fetch of the URI specified by the next attribute of <fetch>
fails for any reason. Previously, the
reason phrase was missing. (ER# 174281799)
If a <merge>
event fails, the connection.merge.failed
event is now generated. Previously, in this scenario, the error.merge
event was generated. (ER# 174561929)
CCP now generates a 400
response when parameter names are
repeated in an HTTP request to I/O processors. (ER# 174281929)
CCP now correctly translates tel URLs (with parameters) to SIP URIs. Previously, the parameter name was not converted to lower case and the parameters needed to be reordered. (ER# 203614207)
CCP now generates an error.semantic
event when it accepts undeclared variables as the sessionid
attribute value
of the <createccxml>
. Previously, this even was not generated. (ER# 174281990)
CCP now sends a SIP BYE
to the connection that failed the merge when the CCXML application
exits. Previously, when a CCXML application used a <merge>
tag, and the endpoint
receiving the SIP REFER
message sent a non-202
SIP
response followed by a SIP NOTIFY
message to the CCP, the merge
failed event was sent to the CCXML application, but the SIP BYE
was not sent when the CCXML application terminated. (ER# 174430601)
The HTTP query string specified as part of the URL is no longer dropped when <fetch>
is executed. (ER# 174281980)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. This section describes new features that were introduced in this release of Voice Platform Call Control Platform (CCP).
This release includes the following corrections and modifications:
CCP now allows the early join to be successful as long as the SDP offer/answer can be negotiated by the time the call is connected. (ER# 209909330)
Supported Operating Systems
New in This Release
Corrections and Modifications
This is a hot fix for this product. This release does not contain new features or functionality.
This release includes the following correction and modification:
The Call Control Platform (CCP) now replies to CANCEL events correctly by sending a SIP 200 OK message, and calls are now terminated properly. Previously, the CCP did not reply correctly to CANCEL events, and calls were not correctly terminated. (ER# 247959409)
Supported Operating Systems
New in This Release
Corrections and Modifications
This release is under shipping control. This section describes new features that were introduced in the initial 8.1 release of Voice Platform Call Control Platform.
This release includes the following corrections and modifications:
Call Control Platform (CCP) now sets connection.disconnected as the reason property when generating the error.conference.join event. Previously, when a BYE message was received from a SIP endpoint while CCP was executing a join operation between that endpoint and another endpoint, the error.conference.join event was generated, but the reason property was not connection.disconnected. (ER# 174430521)
The aai information is now available to the CCXML application when it is passed to CCP as a Request-URI parameter in the SIP INVITE message to CCP. Previously, the aai value was undefined. (ER# 181815975)
A graceful shutdown followed by a stop in Solution Control Interface (SCI) for CCP now cleans up existing calls when the calls are held longer than the CCP option ccpccxml.shutdown_grace_period configured value. (ER# 183796651)
Using hints (hint.confmaxsize) to specify the maximum size for a conference in a <createconference> now works correctly. (ER# 174561985)
This section provides the latest information on known issues and recommendations associated with this product.
When an HTTP client posts an event to the Call Control Platform by using its basicHTTP IO Processor, it must use the exact names for all the headers in the HTTP specification. This limitation is imposed by the basicHTTP IO Processor, which is case sensitive. (ER# 279144735)
Found In: 8.1.301.09 | Fixed In: |
The fetching component of CCP will unexpectedly terminate if it attempts to
fetch an HTTP URL that does not contain a slash in the URL after the http://
.
For example, http://www.genesyslab.com
. (ER# 247567343)
Found In: 8.1.201.92 | Fixed In: 8.1.201.97 |
Call Control Platform has been enhanced to pass non-standard SIP headers from the inbound call to the outbound call belonging to the same CCXML session. The <createcall>
,<dialogprepare>
, and <dialogstart>
tags that create new calls can also pass protocol.sip.headers
hints to add or replace new SIP headers on the outbound call. However, the inbound to outbound SIP header copy will not work if hints that are not of ECMA object type (for example, string) are passed to <createcall>
,<dialogprepare>
,and <dialogstart>
when the outbound call is created. To workaround this, either pass the hints of ECMA Object, or do not pass hints at all. (ER# 244764357)
Found In: 8.1.201.92 | Fixed In: The 8.1.3 release of the Genesys Voice Platform 8.1 User's Guide. |
In rare instances when a <redirect>
fails because the endpoint did not acknowledge the 302 Moved Temporarily
response, the connection.redirected
event is sent to the CCXML application instead of connection.error
event. (ER# 174561923)
Found In: 8.1.201.92 | Fixed In: |
The Call Control Platform currently creates new SIP call legs for conferences and dialogs by using non-secure SIP by default. To force the Call Control Platform to use Secure-SIP (SIPS) for these call legs, configure the [sip]transport.0
and [sip]transport.1
options to disable the TCP and UDP transports. (ER# 223400013)
Found In: 8.1.101.22 | Fixed In: |
The timeout value for <createcall> might not work correctly if the value is less than 32 seconds. If the destination does not respond (for example, a device is down), the timeout will not occur until after 32 seconds have passed. (ER# 230975636)
Found In: 8.1.101.22 | Fixed In: The 8.1.2 release of the Genesys Voice Platform 8.1 CCXML Reference Manual. |
The internal conference memory object may leak if <conferencedestroy> is issued at the same time <disconnect> or <exit/> is issued on the connection/dialog that is currently joined to the conference. To workaround this issue, the CCXML application must be written so that the <conferencedestroy> is issued last, after all of the participants are unjoined from the conference. (ER# 230975146)
Found In: 8.1.101.22 | Fixed In: 8.1.201.92 |
The unjoined-initial-offer-pref value currently does not work for specifying offer-less INVITE versus connection-less-SDP when an initial offer is made to an outbound connection or a dialog. To workaround this issue, modify the offer-less-invite-support and connectionless-sdp-type parameters directly:
For offer-less INVITE:
offer-less-invite-support = true
connectionless-sdp-type = hold or non-routable
For connection-less SDP INVITE:
offer-less-invite-support = false
connectionless-sdp-type = hold or non-routable
(ER# 224262631)
Found In: 8.1.101.22 | Fixed In: 8.1.301.09 |
When the remote end (UAS) sends the offer in 200 OK, the Call Control Platform is forced to provide an answer in ACK. If the offer is invalid and the corresponding answer is not generated, the Call Control Platform does not send an ACK and waits for the SIP transaction timer to clean up the call. To workaround this issue, set the offer-less-invite device profile parameter to false for this device so that the offer/answer occurs between INVITE and 200 OK. (ER# 226950236)
Found In: 8.1.101.22 | Fixed In: 8.1.201.92 |
When an incoming call from the Audio Codes Mediant Gateway has either SRTP or FAX enabled, and lands on either CCP or CTI Connector, the incoming call fails to join to other end points. To workaround this issue, configure the Mediant Gateway to disable both SRTP and FAX in the incoming call. (ER# 227395971, 227446083)
Found In: 8.1.101.22 | Fixed In: 8.1.201.92 |
When the CCXML application invokes a VoiceXML application, the ID of the CCXML connection that invoked the dialog is available through the ccxml.connectionid SIP request-uri parameter of the SIP INVITE invoking the VoiceXML dialog. If <dialogprepare> is executed, this information is not available because of a limitation on the CCP. To workaround this issue, <dialogstart> needs to be called without first executing <dialogprepare>. (ER# 229010775)
Found In: 8.1.101.22 | Fixed In: The 8.1.2 release of the Genesys Voice Platform 8.1 CCXML Reference Manual. |
For backward compatibility reasons with legacy devices, which always require SDP in INVITE, the CCP did not fully conform to RFC 3264. In particular, if MCP needs to perform a Media Redirect Transfer with CCP, the mrt.sendsdpininvite parameter in MCP Sessmgr configuration section will have to be enabled. Additionally, the descriptions in the Genesys Voice Platform 8.1 User's Guide should be updated as follows for the following parameters:
In the Sessmgr configuration section, for the mrt.sendsdpininvite parameter:
NGI application—When enabled, for a Media Redirect call that has connectwhen specified as answered, MCP will send the caller's last SDP to the called party both in reINVITE and in ACK. When disabled, the reINVITE to the called party will not contain any SDP.
GVPi application—When enabled, for a Media Redirect call, MCP will send the caller's last SDP to the called party both in reINVITE and in ACK in the case of 2signal channel . Disable it so that reINVITE to the called party will not contain any SDP.
(ER# 209909330)
Found In: 8.1.001.60 | Fixed In: 8.1.101.22 |
Call Control Platform (CCP) incorrectly translates tel URLs (with parameters) to SIP URIs. The parameter name is not converted to lower case and the parameters need to be reordered. (ER# 203614207)
Found In: 8.1.001.60 | Fixed In: 8.1.201.92 |
The error.conference.create event contains an incorrect attribute value. The event will contain the following attributes:
However, the conferenceid and eventsource attributes will not be set to the conference ID. The eventsource will be set to the CCXML Session ID that triggered the error.conference.create event. This is because the conference ID is being generated after the device profile selection and the only way the error.conference.create event can be triggered is to not set a default Device Profile for Default Conference. Therefore, the error.conference.create event will be generated before reaching the point where the conference ID is generated.
(ER# 174430617)
Found In: 8.1.001.60 | Fixed In: The 8.1.2 release of the Genesys Voice Platform 8.1 CCXML Reference Manual. |
When a CCXML application uses a <merge>
tag, and the endpoint
receiving the SIP REFER
message sends a non-202
SIP
response followed by a SIP NOTIFY
message to the CCP, the merge
failed event is sent to the CCXML application. However, if the CCXML application
exits, a SIP BYE
is not sent to the endpoint. (ER# 174430601)
Found In: 8.0.004.47 | Fixed In: 8.1.201.92 |
If a <merge>
event fails, the error.merge
event is generated. This conforms to the 2005/06/29 Last Call Working Draft
CCXML specification which specifies that the error.merge
event
is the correct event. The latest CCXML specification specifies that a connection.merge.failed
should be thrown for failed merges. (ER# 174561929)
Found In: 8.0.004.47 | Fixed In: 8.1.201.92 |
CCP does not send two connection.merged
events for each connection
if the <disconnect>
event occurs immediately after the
<merge>
event within the same transition. Instead, CCP
sends a connection.merged
event and a connection.disconnected
event to the application. (ER# 183951622)
Found In: 8.0.004.47 | Fixed In: 8.1.400.52 |
CCP incorrectly sends the connection.redirected
event to the CCXML
application instead of the connection.error
event when a <redirect>
fails because the endpoint does not acknowledge (ACK
) the 302
Moved Temporarily
response. (ER# 174561923, 173514335)
Found In: 8.0.004.47 | Fixed In: |
If an initial bridge (A<->B) is overridden by a CCXML join (join A<-B),
and one of the endpoints (B) disconnects in the middle of receiving a re-INVITE
in order to establish the new bridge, CCP does not send the conference.unjoined
event for the initial bridge (A<->B). However, CCP sends the connection.disconnected
(for connection B) and error.conference.join
(for join A<-B)
correctly. (ER# 174561941)
Found In: 8.0.004.47 | Fixed In: 8.0.401.03 |
CCP accepts undeclared variables as the sessionid
attribute value
of the <createccxml>
element without generating the error.semantic
event. (ER# 174281990)
Found In: 8.0.004.47 | Fixed In: 8.1.201.92 |
CCP does not generate a 400
response when parameter names are
repeated in an HTTP request to I/O processors. (ER# 174281929)
Found In: 8.0.004.47 | Fixed In: 8.1.201.92 |
If the fetch of the URI specified by the next attribute of <fetch>
fails for any reason, the error.fetch
event is generated. If the
URI has a scheme of http:
, the reason property of the event should
read: Fetch failed: <error code> <reason phrase>
,
where <error code>
is the HTTP error code and <reason
phrase>
is the HTTP reason phrase in the response. Currently, the
reason phrase is missing. (ER# 174281799)
Found In: 8.0.004.47 | Fixed In: 8.1.201.92 |
When using Offer-Answer
, an incoming connection cannot be joined
to two outbound connections if the alerting connection is accepted in a later
transition than the joins. (ER# 174430561)
Found In: 8.0.004.47 | Fixed In: |
The src
attribute is dropped when passing a query string of the
original URL of the next
element.
For example, in the following <fetch>
tag:
<fetch next="'next.jsp?t=5'" namelist="a b"/>
Parameters a
and b
are passed through the resulting query string: next.jsp?a=1&b=2
. The parameter t
should also be passed through the query string. (ER# 174281980)
Found In: 8.0.004.47 | Fixed In: 8.1.201.92 |
CCP allows the use of the <move>
tag when the destination
session has not been created when using the <createccxml>
tag. (ER# 174282000)
Found In: 8.0.004.47 | Fixed In: |
This section documents features that are no longer supported in this software. This cumulative list is in release-number order with the most recently discontinued features at the top of the list.
There are no discontinued items for this release.
Information in this section is included for international customers.
There are no known internationalization issues for this product.
Additional information on Genesys Telecommunications Laboratories, Inc. is available on our Customer Care website. The following documentation also contains information about this software. Please consult the Genesys Voice Platform 8.1 Deployment Guide first.
Genesys Voice Platform 8.1 Deployment Guide, which provides information about installing and configuring Genesys Voice Platform (GVP).
Genesys Voice Platform 8.1 User's Guide, which provides information about configuring, provisioning, and monitoring GVP and its components.
Genesys Voice Platform 8.1 Genesys VoiceXML 2.1 Reference Help, which provides information about developing Voice Extensible Markup Language (VoiceXML) applications. It presents VoiceXML concepts, and provides examples that focus on the GVP Next Generation Interpreter (NGI) implementation of VoiceXML.
Genesys Voice Platform 8.1 Legacy Genesys VoiceXML 2.1 Reference Manual, which describes the VoiceXML 2.1 language as implemented by the Legacy GVP Interpreter (GVPi) in GVP 7.6 and earlier, and which is now supported in the GVP 8.1 release.
Genesys Voice Platform 8.1 CCXML Reference Manual, which provides information about developing Call Control Extensible Markup Language (CCXML) applications for GVP.
Genesys Voice Platform 8.1 Troubleshooting Guide, which provides troubleshooting methodology, basic troubleshooting information, and troubleshooting tools.
Genesys Voice Platform 8.1 Configuration Options Reference, which replicates the metadata available in the Genesys provisioning GUI, to provide information about all the GVP configuration options, including descriptions, syntax, valid values, and default values.
Genesys Voice Platform 8.1 Metrics Reference, which provides information about all the GVP metrics (VoiceXML and CCXML application event logs), including descriptions, format, logging level, source component, and metric ID.
Genesys Voice Platform 8.1 SNMP and MIB Reference, which provides information about all of the Simple Network Management Protocol (SNMP) Management Information Bases (MIBs) and traps for GVP, including descriptions and user actions.
Genesys Voice Platform 8.1 Web Services API wiki, which describes the Web Services API that the Reporting Server supports.
Voice Platform Solution 8.1 Integration Guide, which provides information about integrating GVP 8.1, SIP Server 8.0, and, if applicable, IVR Server.
Composer 8.1 Deployment Guide, which provides installation and configuration instructions for Composer.
Composer 8.1 Help, which provides online information about using Composer, an Integrated Development Environment used to develop applications for Genesys Voice Platform and Universal Routing.
Framework 8.1 Deployment Guide, which provides information about configuring, installing, starting, and stopping Framework components.
Framework 8.1 Genesys Administrator Help, which provides information about configuring and provisioning contact center objects by using the Genesys Administrator.
Framework 8.1 Genesys Administrator Deployment Guide, which provides information about installing and configuring Genesys Administrator.
Framework 8.1 Configuration Options Reference Manual, which provides descriptions of the configuration options for Framework components.
Framework 8.1 SIP Server Deployment Guide, which provides information about configuring and installing SIP Server.
Product documentation is provided on the Customer Care website, the Genesys Documentation website, and the Documentation Library DVD (produced monthly) or the Developer Documentation CD.
Note: For the DVD/CD, the New Documents on this DVD/CD page indicates the production date for that disc. Due to disc production schedules, documentation on the Genesys Documentation website may be more up-to-date than what is available on disc immediately after a product is released or updated. To determine the version of a document, check the version number that is located on the second page in PDFs or on the About This File topic in Help files.