PEC-GS/Current/Administrator/SPOptions - Revision history

WikiSysop at 16:30, September 28, 2023

2023-09-28T16:30:58Z

WikiSysop at 19:46, September 12, 2022

2022-09-12T19:46:05Z

WikiSysop at 17:30, July 20, 2022

2022-07-20T17:30:52Z

Brian.marshall@genesys.com: Published

2021-12-13T19:12:17Z

Published

Brian.marshall@genesys.com: Published

2021-12-08T16:38:19Z

Published

WikiSysop: Text replacement - "\|Platform=([^\|])GenesysEngage-onpremises([\|])" to "|Platform=$1GenesysEngage-cloud$2"

2020-07-25T02:32:07Z

Text replacement - "\|Platform=([^\|]*)GenesysEngage-onpremises([\|]*)" to "|Platform=$1GenesysEngage-cloud$2"

WikiSysop: Text replacement - "\|Platforms?=([^\|])PureEngage([\|])" to "|Platform=$1GenesysEngage-onpremises$2"

2020-07-15T11:44:50Z

Text replacement - "\|Platforms?=([^\|]*)PureEngage([\|]*)" to "|Platform=$1GenesysEngage-onpremises$2"

Mwest: Published

2020-06-09T19:06:38Z

Published

Bgrenon: Published

2020-06-08T14:31:58Z

Published

Show changes

@@ Line 67: / Line 67: @@
 {{!}}The first user's DN extension as configured in the configuration database. Included in the SIP URI&mdash;for example, <sip:'''DN0'''@serverHostName0:port0>
 {{!}}-
 {{!}}server
 {{!}}
 {{!}}The SIP Server or Proxy location for the first user. Included in the SIP URI&mdash;for example, <sip:DN0@'''serverHostName0:port0'''>
 {{!}}-
 {{!}}protocol
 {{!}}
@@ Line 149: / Line 153: @@
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=deferdevicerelease|display text=defer_device_release}}
 {{!}}-
-{{!}} rowspan="25"{{!}}
+{{!}} rowspan="26"{{!}}
 {{!}} colspan="3"{{!}}'''session'''
 {{!}}-
-{{!}} rowspan="24"{{!}}
+{{!}} rowspan="25"{{!}}
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=agc_mode|display text=agc_mode}}
 {{!}}-
@@ Line 190: / Line 194: @@
 {{!}}-
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=ringing_file|display text=ringing_file}}
 {{!}}-
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=restart_audio_if_stuck|display text=restart_audio_if_stuck}}
@@ Line 350: / Line 356: @@
 *IPv4—the application selects an available local IPv4 address; IPv6 addresses are ignored.
 *IPv6—the application selects an available local IPv6 address; IPv4 addresses are ignored.
-*IPv4,IPv6 or an empty—the application selects an IPv4 address if one exists. If not, an available IPv6 address is selected.
+*IPv4, IPv6 or an empty—the application selects an IPv4 address if one exists. If not, an available IPv6 address is selected.
-*IPv6,IPv4—the application selects an IPv6 address if one exists. If not, an available IPv4 address is selected.
+*IPv6, IPv4—the application selects an IPv6 address if one exists. If not, an available IPv4 address is selected.
 '''Note:''' This parameter has no effect if the '''public_address''' option specifies an explicit IP address.
@@ Line 725: / Line 731: @@
 Uncompressed PCM audio must 16 bit mono or stereo and have a frequency of 8, 16, or 32 KHZ.
 {{AnchorDiv|restart_audio_if_stuck}}

@@ Line 24: / Line 24: @@
 |anchor=BasicContainer
 |alignment=Vertical
-|structuredtext=The first container ("Basic") holds the basic connectivity details that are required to connect to your SIP Server. This container has at least one connection (Connectivity) element with the following attributes:
+|structuredtext=The first container ("Basic") holds the basic connectivity details that are required to connect to your SIP Server, optionally through a Session Border Controller (SBC). This container has at least one connection (Connectivity) element with the following attributes:
   &lt;Connectivity user="DN" server="SERVER:PORT" protocol="TRANSPORT"/&gt;
@@ Line 33: / Line 33: @@
 *<tt>user="DN"</tt>: Supply a valid DN for the user attribute.
-*<tt>server="SERVER:PORT"</tt>: Replace SERVER with the host name where your SIP Server is deployed, and PORT with the SIP port of the SIP Server host. The default SIP port value is 5060. For SRV resolution, specify the SRV record without including the port number in the server's URI. Also see {{Link-SomewhereInThisManual|topic=SPOptions|anchor=SRV_resolution|display text=SRV Resolution}} below.
+*<tt>server="SERVER:PORT"</tt>: Replace SERVER with the host name where your SIP Server or SBC is deployed, and PORT with the SIP port of the SIP Server or SBC host. The default SIP port value is 5060. For SRV resolution, specify the SRV record without including the port number in the server's URI. Also see {{Link-SomewhereInThisManual|topic=SPOptions|anchor=SRV_resolution|display text=SRV Resolution}} below.
-*<tt>protocol="TRANSPORT"</tt>: Set the protocol attribute to reflect the protocol being used to communicate with SIP Server. Possible values are <tt>UDP</tt>, <tt>TCP</tt>, or <tt>TLS</tt>.
+*<tt>protocol="TRANSPORT"</tt>: Set the protocol attribute to reflect the protocol being used to communicate with SIP Server or SBC. Possible values are <tt>UDP</tt>, <tt>TCP</tt>, or <tt>TLS</tt>.
 {{AnchorDiv|SRV_resolution}}
 ===SRV resolution===

@@ Line 61: / Line 61: @@
 !'''Description'''
 {{!}}-
-{{!}} rowspan="" 4{{!}}
+{{!}} rowspan="" 3{{!}}
-{{!}} rowspan="" 4{{!}}Connectivity
+{{!}} rowspan="" 3{{!}}Connectivity
 {{!}}user
 {{!}}
@@ Line 149: / Line 149: @@
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=deferdevicerelease|display text=defer_device_release}}
 {{!}}-
-{{!}} rowspan="23"{{!}}
+{{!}} rowspan="25"{{!}}
 {{!}} colspan="3"{{!}}'''session'''
 {{!}}-
-{{!}} rowspan="22"{{!}}
+{{!}} rowspan="24"{{!}}
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=agc_mode|display text=agc_mode}}
 {{!}}-
@@ Line 160: / Line 160: @@
 {{!}}-
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=auto_answer_delay|display text=auto_answer_delay}}
 {{!}}-
 {{!}}{{Link-SomewhereInThisVersion|manual=Administrator|topic=SPOptions|anchor=dtmf_feedback|display text=dtmf_feedback}}
@@ Line 197: / Line 201: @@
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=rx_agc_mode|display text=rx_agc_mode}}
 {{!}}-
-{{!}} rowspan="6"{{!}}
+{{!}} rowspan="7"{{!}}
 {{!}} colspan="3"{{!}}'''device'''
 {{!}}-
-{{!}} rowspan="5"{{!}}
+{{!}} rowspan="6"{{!}}
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=audio_in_device|display text=audio_in_device}}
 {{!}}-
@@ Line 208: / Line 212: @@
 {{!}}-
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=headset_name|display text=headset_name}}
 {{!}}-
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=use_headset|display text=use_headset}}
@@ Line 564: / Line 570: @@
 Time in milliseconds to wait before auto-answering. The recommended and default value is 1500 milliseconds.
 {{AnchorDiv|dtmf_feedback}}
@@ Line 748: / Line 778: @@
 {{AnchorDiv|audio_out_device}}
 ====audio_out_device====
 Valid Values: A regex that matches the [http://www.cplusplus.com/reference/regex/ECMAScript/ ECMAScript] standard.
@@ Line 765: / Line 794: @@
 The name of the headset model.
 {{AnchorDiv|ringer_device}}

@@ Line 149: / Line 149: @@
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=deferdevicerelease|display text=defer_device_release}}
 {{!}}-
-{{!}} rowspan="22"{{!}}
+{{!}} rowspan="23"{{!}}
 {{!}} colspan="3"{{!}}'''session'''
 {{!}}-
-{{!}} rowspan="21"{{!}}
+{{!}} rowspan="22"{{!}}
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=agc_mode|display text=agc_mode}}
 {{!}}-
@@ Line 160: / Line 160: @@
 {{!}}-
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=auto_answer_delay|display text=auto_answer_delay}}
 {{!}}-
 {{!}}{{!}}{{Link-SomewhereInThisManual|topic=SPOptions|anchor=dtmf_method|display text=dtmf_method}}
@@ Line 562: / Line 564: @@
 Time in milliseconds to wait before auto-answering. The recommended and default value is 1500 milliseconds.
 {{AnchorDiv|dtmf_method}}

@@ Line 7: / Line 7: @@
 |Platform=GenesysEngage-cloud
 |Role=Administrator
 |Section={{Section
 |alignment=Vertical
 |structuredtext=For an example of the configuration file, see {{Link-SomewhereInThisVersion|manual=Administrator|topic=SPDeploy|anchor=configuring|display text=Configuring Genesys Softphone}}.
@@ Line 24: / Line 19: @@
 #Repackage Genesys Softphone with the custom '''Softphone.config''' file through an IT-controlled installation.
 #Push the custom package to the agent workstations.
 |Status=No
 }}{{Section
@@ Line 31: / Line 24: @@
 |anchor=BasicContainer
 |alignment=Vertical
 |structuredtext=The first container ("Basic") holds the basic connectivity details that are required to connect to your SIP Server. This container has at least one connection (Connectivity) element with the following attributes:
@@ Line 70: / Line 61: @@
 !'''Description'''
 {{!}}-
-{{!}} rowspan=4{{!}}
+{{!}} rowspan="" 4{{!}}
-{{!}} rowspan=4{{!}} Connectivity
+{{!}} rowspan="" 4{{!}}Connectivity
-{{!}} user
+{{!}}user
 {{!}}
 {{!}}The first user's DN extension as configured in the configuration database. Included in the SIP URI&mdash;for example, <sip:'''DN0'''@serverHostName0:port0>
@@ Line 85: / Line 76: @@
 {{!}}-
 {{!}}}
 |Status=No
 }}{{Section
@@ Line 92: / Line 81: @@
 |anchor=GenesysContainer
 |alignment=Vertical
 |structuredtext=The second Container ("Genesys") holds a number of configurable settings that are organized into domains and sections. You don't have to change these settings but you can customize them.
@@ Line 318: / Line 305: @@
 {{!}}{{!}} {{Link-SomewhereInThisManual|topic=SPOptions|anchor=ringing_file|display text=ringing file}}
 {{!}}}
 |Status=No
 }}{{Section
@@ Line 325: / Line 310: @@
 |anchor=policy
 |alignment=Vertical
 |structuredtext={{AnchorDiv|enpoint}}
 ==='''endpoint''' Section===
@@ Line 788: / Line 771: @@
 If set to <tt>1</tt>, the SDK uses a headset as the preferred audio input and output device and the audio devices specified in audio_in_device and audio_out_device are ignored.
-|structuredtextwide=
-|FAQHeading=
+{{AnchorDiv|exclude_headset}}
 |Status=No
 }}{{Section
@@ Line 795: / Line 786: @@
 |anchor=codecs
 |alignment=Vertical
 |structuredtext=See [https://docs.genesys.com/Documentation:SESDK:Developer:ConfiguringforNET:9.0.0NET#Working_with_Codec_Priorities SIP Endpoint SDK for .NET 9.0.0NET&mdash;Working with Codec Priorities].
 |Status=No
 }}{{Section
@@ Line 805: / Line 792: @@
 |anchor=proxies
 |alignment=Vertical
 |structuredtext=Configure a proxy section for each connectivity line. For example, for three connectivity lines, configure sections for <tt>proxy0</tt>, <tt>proxy1</tt>, and <tt>proxy2</tt>.
@@ Line 960: / Line 945: @@
 User ID for TURN authorization{{NoteFormat|Starting from 9.0.012.02, this setting is deprecated and is not recommended for use, unless suggested by Genesys Technical Support to fix specific problems.
 Use the GCTI_TURN_USERNAME environment variable to set the username for TURN authentication.|3}}
 |Status=No
 }}{{Section
@@ Line 967: / Line 950: @@
 |anchor=System
 |alignment=Vertical
 |structuredtext=<span id="sdk_system" data-origid="sdk_system"></span>
@@ Line 1,129: / Line 1,110: @@
 The Ringing sound file name in the current directory or the full local path to the ringing sound file.
 |Status=No
 }}
 }}

@@ Line 5: / Line 5: @@
 |Context=Configuration settings by container and domain found in the Softphone.config file in the Genesys Softphone Installation Directory.
 |ComingSoon=No
-|Platform=GenesysEngage-onpremises
+|Platform=GenesysEngage-cloud
 |Role=Administrator
 |Prereq=

PEC-GS/Current/Administrator/SPOptions - Revision history

WikiSysop at 16:30, September 28, 2023

WikiSysop at 19:46, September 12, 2022

WikiSysop at 17:30, July 20, 2022

Brian.marshall@genesys.com: Published

Brian.marshall@genesys.com: Published

WikiSysop: Text replacement - "\|Platform=([^\|]*)GenesysEngage-onpremises([\|]*)" to "|Platform=$1GenesysEngage-cloud$2"

WikiSysop: Text replacement - "\|Platforms?=([^\|]*)PureEngage([\|]*)" to "|Platform=$1GenesysEngage-onpremises$2"

Mwest: Published

Bgrenon: Published

WikiSysop: Text replacement - "\|Platform=([^\|])GenesysEngage-onpremises([\|])" to "|Platform=$1GenesysEngage-cloud$2"

WikiSysop: Text replacement - "\|Platforms?=([^\|])PureEngage([\|])" to "|Platform=$1GenesysEngage-onpremises$2"