When you install the Adobe Connect Server, all of the required files for the telephony adaptors install. They are installed whether you select them in the installation wizard or not. If you do not select a particular telephony adaptor, the configuration screen for the adaptor doesn't appear. Therefore, you cannot use the adaptor with Adobe Connect until manually configuring it.  To configure the telephony adaptors, edit the telephony-settings.xml file, using the steps below. Follow only the configuration steps that apply to your specific telephony provider. 

  1. In a text editor, open the [root_install_dir]\TelephonyService\conf\telephony-settings.xml. 
  2. In the XML file, define the dial-in sequence to the audio conferencing provider. 
  3. Validate and save the XML file. 
  4. Restart Adobe Connect Central Application Server.

You provide values for the parameters in brackets ([]). The adaptor provides values for the parameters in curly brackets ({}). 

The following sample uses the dial-in sequence for the PGi adaptor:

<telephony-adaptor id="premiere-adaptor" class-name="com.macromedia.breeze_ext.premiere.gateway.PTekGateway" enabled="true" name="{premiere-adaptor}" 
disable-profiles-on-edit="false" disable-profiles-on-disable="false">
    
    <setting id="PREMIERE_HOST">CSAXIS.PREMCONF.COM </setting> 
    
    <setting id="PREMIERE_PORT">443</setting> 

    <setting id="PREMIERE_WEB_ID">[123456]</setting> 

    <setting id="PREMIERE_PASSWORD">[aVerySecurePassword]</setting> 

    <setting id="PREMIERE_MAX_DOWNLOAD_TRIES">120</setting> 

    <setting id="PREMIERE_DOWNLOAD_LOGIN">[login]</setting> 

    <setting id="PREMIERE_DOWNLOAD_PASSWORD">[aVerySecurePassword]</setting> 

    <setting id="PREMIERE_REPORT_INTERVAL">60</setting> 

    <setting id="PREMIERE_DOWNLOAD_URL">https://ww7.premconf.com/audio/</setting> 

    <dial-in-sequence> 

        <conf-num>{x-tel-premiere-conference-number}</conf-num> 

        <delay>6000</delay> 

        <dtmf>{x-tel-premiere-participant-code}</dtmf> 

        <dtmf>#</dtmf> 

        <delay>2000</delay> 

        <dtmf>*#</dtmf> 

        <delay>5000</delay> 

    </dial-in-sequence> 

</telephony-adaptor>

 

Basic Premiere adaptor settings

Adaptor Setting Required Description
PGi NA Adaptor/PGi EMEA Adaptor PREMIERE_HOST/ EMEA_PREMIERE_HOST Yes The host name or IP address of the PGi server. Get value from PGi Global Services.
PGi NA Adaptor/PGi EMEA Adaptor PREMIERE_PORT/ EMEA_PREMIERE_PORT Yes The port that Connect Pro uses to connect to the PGi server. Get value from PGi Global Services.
PGi NA Adaptor/PGi EMEA Adaptor PREMIERE_WEB_ID/ EMEA_PREMIERE_WEB_ID Yes The ID used to access the PGi bridge from the adaptor. Get value from PGi Global Services.
PGi NA Adaptor/PGi EMEA Adaptor PREMIERE_PASSWORD/ EMEA_PREMIERE_PASSWORD Yes The password used to connect to the PGi API. Get value from PGi Global Services.
PGi NA Adaptor/PGi EMEA Adaptor PREMIERE_DOWNLOAD_URL / EMEA_PREMIERE_ DOWNLOAD_URL No For NA - The URL that Connect Pro uses to download recordings from the PGi audio conference service. The default value is https://ww5.premconf.com/audio/. For EMEA - The URL that Connect Pro uses to download recordings from the PGi audio conference service. The default value is http://eurecordings.premiereglobal.ie/audio/.
PGi NA Adaptor/PGi EMEA Adaptor PREMIERE_DOWNLOAD_ LOGIN / EMEA_PREMIERE_ DOWNLOAD_LOGIN Yes The login used to download PGi audio recordings. This value is the same as the value of PREMIERE_WEB_ID.
PGi NA Adaptor/PGi EMEA Adaptor PREMIERE_DOWNLOAD_ PASSWORD/ EMEA_PREMIERE_ DOWNLOAD_PASSWORD Yes The password used to download PGi audio recordings. This value is the same as the value of PREMIERE_PASSWORD.

Advanced  Premiere adaptor settings

Adapter Setting Required Default Value Description
PGi NA Adaptor PREMIERE_ MAX_ DOWNLOAD_ TRIES No 1440 The maximum number of times Connect tries to download audio recordings from PGi.
PGi NA Adaptor PREMIERE_ REPORT_ INTERVAL No 600,000 (10 minutes) The time interval at which to print a report of audio downloads, in milliseconds. This audio download report is printed in the PGi log file.
PGi NA Adaptor PREMIERE_ PING_ INTERVAL No 60,000 (1 minute) Interval at which the scheduler does a sweep, in milliseconds. The scheduled sweep takes care of attempting pending downloads, cleans up internal data structures, and closes unused sockets.
PGi NA Adaptor PREMIERE_ TELEPHONY_ TIMEOUT No 10,000 (10 seconds) Length of time for which adaptor waits for a response from the PGi bridge after sending a request, in milliseconds. If the response is not received within the specified number of milliseconds, an exception is raised.
PGi NA Adaptor PREMIERE_ MAX_ CONNECTION_ TRIES No 3 Maximum number of allowed attempts to connect to PGi bridge before sending a message that the connection failed. 
PGi NA Adaptor PREMIERE_ EMEA No no Boolean value that specifies whether the PGi adaptor you have installed is EMEA (yes) or NA (no). Setting this value to yes indicates that this adaptor is used with a PGi EMEA bridge. 
PGi NA Adaptor PREMIERE_ DEBUG_ ALL No yes Boolean value that specifies whether debugging is enabled. If debugging is enabled, verbose logging is stored in the PGi adaptor logs.
PGi NA Adaptor PREMIERE_ TALKER_ MESSAGE_ THROTTLE_ THRESHOLD No 1,000 (one second) Interval at which the telephony service sends talker messages to a Connect meeting, in milliseconds. (Talker messages cause the speaking icon to show  for an attendee in the meeting).  Set to 0 or -1 to disable throttling. A low value can result in many talker messages, which increases the load on the server. Using a large value can result in a significant lag between the time someone starts speaking and when the icon is updated.
PGi NA Adaptor PREMIERE_ DTMF_ PREFIX_ TOKEN No *29 Characters that indicate the DTMF entry is a token. Change this value only if the value has been changed on the bridge.
PGi NA Adaptor PREMIERE_ TOKEN_ LENGTH No 4 Number of digits in the unique token Connect generates for each user attending a meeting. 
PGi NA Adaptor PREMIERE_ DTMF_ POSTFIX_ TOKEN No #* Characters that indicate the token is completed, which signals Connect  to generate the token to merge the phone user with a web user. Change this value only if the value has been changed on the bridge.
PGi EMEA Adaptor EMEA_ PREMIERE_ MAX_ DOWNLOAD_ TRIES No 1440 The maximum number of times Connect Pro tries to download audio recordings from PGi.
PGi EMEA Adaptor EMEA_ PREMIERE_ REPORT_ INTERVAL No 600,000 (10 minutes) Interval at which to print a report of audio downloads, in milliseconds. This audio download report is printed in the PGi log file.
PGi EMEA Adaptor EMEA_ PREMIERE_ PING_ INTERVAL No 60,000 (1 minute) Interval at which the scheduler does a sweep, in milliseconds. The scheduled sweep takes care of attempting pending downloads, cleans up internal data structures, and closes unused sockets.
PGi EMEA Adaptor EMEA_ PREMIERE_ TELEPHONY_ TIMEOUT  No 10,000 (10 seconds) Length of time for which adaptor waits for a response from the PGi bridge after sending a request, in milliseconds. If the response is not received within the specified number of milliseconds, an exception is raised.
PGi EMEA Adaptor EMEA_ PREMIERE_ MAX_ CONNECTION_ TRIES No 3 Maximum number of allowed attempts to connect to PGi bridge in case of failure. 
PGi EMEA Adaptor EMEA_ PREMIERE_ UV_ COUNTRY_ CODE No UK, The code for the country whose number to use for the universal line for dialing in. Preferably set it to the location where your service provider is located.
PGi EMEA Adaptor EMEA_ PREMIERE_ DEBUG_ ALL No yes Boolean value that specifies whether debugging is enabled. If debugging is enabled, verbose logging is stored in the PGi adaptor logs.
PGi EMEA Adaptor EMEA_ PREMIERE_ TALKER_ MESSAGE_ THROTTLE_ THRESHOLD No 1,000 (one second) Interval at which the telephony service sends talker messages to a Connect meeting, in milliseconds. (Talker messages cause the speaking icon to show for an attendee in the meeting).  Set to 0 or -1 to disable throttling.  Setting this to a low value can result in large number of talker messages and increases load on server. Using a large value can cause speaking icon to update after long time intervals.
PGi EMEA Adaptor EMEA_ PREMIERE_ MAX_ NUMBER_ FIELDS No 6 Maximum number of fields that are created in the database, to store the list of conference numbers. The bridge returns the conference numbers, are chunks them into as many fields as specified.
PGi EMEA Adaptor EMEA_ PREMIERE_ MAX_ NUMBERS No 18 Maximum number of phone numbers allowed to be stored in Connect.
PGi EMEA Adaptor EMEA_ PREMIERE_ EMEA_ NUMBER_ MARKER No _ (underline) Character that identifies which conference numbers that the audio bridge returns are  EMEA conference numbers.
PGi EMEA Adaptor EMEA_ PREMIERE_ EMEA_ NUMBER_ ALG_ FALLBACK No no Boolean value that specifies whether to use the fallback algorithm for storing conference numbers. Using the fallback algorithm ensures that the EMEA number marker is ignored when choosing the conference numbers to store in Connect.
PGi EMEA Adaptor EMEA_ PREMIERE_ ALLOW_ TOLL_ UV_ NUMBER No no Boolean value that specifies whether a toll conference number can be used for Universal Line. Setting this value to yes ensures that the adaptor uses a toll number  when it's unable to get a toll-free number for the specified country. Countries as specified in "EMEA_PREMIERE_UV_COUNTRY_CODE" setting
PGi EMEA Adaptor PREMIERE_ DTMF_ PREFIX_ TOKEN No *29 Characters that indicate the DTMF entry is a token. Change this value only if the value has been changed on the bridge.
PGi EMEA Adaptor PREMIERE_ TOKEN_ LENGTH No 4 Number of digits in the unique token Connect generates for each user attending a meeting. 
PGi EMEA Adaptor PREMIERE_ DTMF_ POSTFIX_ TOKEN No #* Characters that indicate the token is completed, which signals Connect  to generate the token to merge the phone user with a web user. Change this value only if the value has been changed on the bridge.

 The following sample uses the dial-in sequence for the InterCall adaptor:

<telephony-adaptor id="intercall-adaptor" class-name="com.macromedia.breeze_ext.telephony.Intercall.IntercallTelephonyAdaptor" 
enabled="true" name="{intercall-adaptor}" disable-profiles-on-edit="false" disable-profiles-on-disable="false" default-recording-source="audio-bridge"> 

    <setting id="INTERCALL_CCAPI_HOST">https://iccapi.audiocontrols.net:8443/axis2/services/CCAPI</setting> 

    <setting id="INTERCALL_CCAPI_AUTH_HOST">https://iccapi.audiocontrols.net:8443/axis2/services/Authorization</setting> 

    <setting id="INTERCALL_CLIENT_CALLBACK_URL">https://[external-hostname]:8443/services/CCAPICallbackSOAP</setting> 

    <setting id="INTERCALL_APP_TOKEN">[appTokenProvidedByIntercall]</setting> 

    <setting id="INTERCALL_BREEZE_INSTALL">C:\breeze</setting> 

    <dial-in-sequence>         <conf-num>{x-tel-intercall-conference-number}</conf-num>         <delay>6000</delay>         <dtmf>{x-tel-intercall-participant-code}</dtmf>         <dtmf>#</dtmf>         <delay>4000</delay>         <dtmf>#</dtmf>         <delay>8000</delay>         <dtmf>#</dtmf> 

    </dial-in-sequence> 

</telephony-adaptor>

 Basic InterCall adaptor settings

Adaptor Setting Required Description
InterCall Adaptor INTERCALL_CCAPI_HOST Yes The host URL for the InterCall CCAPI service.
InterCall Adaptor INTERCALL_CCAPI_AUTH_HOST Yes The host  URL for the InterCall CCAPI Authorization service.
InterCall Adaptor INTERCALL_CLIENT_CALLBACK_URL Yes The callback URL of Connect for InterCall to callback.
InterCall Adaptor INTERCALL_APP_TOKEN Yes The app token used for getting the service provider instance from the bridge.
InterCall Adaptor INTERCALL_EMEA_COUNTRY_CODES Yes The country codes for which the conference numbers are displayed. For example, UK; FR; DE; IT; ES; AU; AT; BE; CN; IN; IE; IT; JP; RU; CH; US

Advanced InterCall adaptor settings 

Adaptor Setting Required Default Value Description
InterCall Adaptor INTERCALL_HEARTBEAT_INTERVAL No 60,000 (1 minute) The time interval in milliseconds for sending conversation heartbeats to the bridge. Sending heartbeats to InterCall bridge is necessary to keep a session alive on it. This Interval must not be more than 2 minutes. 
InterCall Adaptor INTERCALL_DEBUG No FALSE Indicates if the adaptor is to run in debug mode which results in verbose logging in the InterCall adaptor logs.
InterCall Adaptor INTERCALL_ACTIVE_SCO_TEST_INTERVAL No  10 Specifies the number of heartbeats to skip before checking for the activeness of a meeting in connect. This setting ensures that sessions do not continue to linger forever and are checked for activeness after specified number of heartbeats.
InterCall Adaptor INTERCALL_DTMF_PREFIX_TOKEN No  #1 Characters that indicate the DTMF entry is a token. Change this value only if the value has been changed on the bridge.
InterCall Adaptor INTERCALL_TOKEN_LENGTH No  4 Number of digits in the unique token Connect generates for each user attending a meeting. 
InterCall Adaptor INTERCALL_DTMF_POSTFIX_TOKEN No  # Characters that indicate the token is completed, which signals Connect  to generate the token to merge the phone user with a web user. Change this value only if the value has been changed on the bridge.
InterCall Adaptor INTERCALL_EMEA_DIALIN_NUMBER_TYPES No    The dial-in conference number types to get from InterCall for storing in Connect. Contact InterCall for the number types. Suggested values are IT; NF, and so on.
InterCall Adaptor INTERCALL_TOLL_FREE_COUNTRY_CODE No  US Represents the code for the country whose number is used for the Universal line to dial in. Preferably set it to the location where you service provider is located.

The following sample uses the dial-in sequence for the Avaya adaptor:

<telephony-adaptor id="avaya-adaptor" class-name="com.macromedia.breeze_ext.telephony.AvayaAdaptor" enabled="true" disable-profiles-on-disable="true" disable-profiles-on-edit="true">
    <setting id="AVAYA_AUDIO_CONVERTER_PATH">${app.root}/util/avaya/</setting>
    <setting id="AVAYA_BRIDGE_NAME">10.59.72.95</setting>
    <setting id="DISABLE_DIALOUT">false</setting>
    <setting id="AVAYA_DTMF_CAPTURE_MAX_TIMEOUT">10</setting>
    <setting id="AVAYA_DTMF_CAPTURE_MAX_LENGTH">20</setting>
    <setting id="AVAYA_FTP_DIRECTORY">/usr3/confrp/</setting>
    <setting id="AVAYA_FILE_TRANSFER">autodetect</setting>
    <setting id="AVAYA_FTP_LOGIN">dcbguest</setting>
    <setting id="AVAYA_FTP_PASSWORD">#E&amp;P!W#D%pgg8caizyfX6WNQfngwpJQ==</setting>
    <setting id="AVAYA_LOGINID">op1</setting>
    <setting id="AVAYA_PASSWORD">#E&amp;P!W#D%TD2qU+1+gYE=</setting>
    <setting id="AVAYA_MUSICSOURCE">1</setting>
    <setting id="AVAYA_PHONEOPERATOR_ID">0</setting>
    <setting id="AVAYA_VALIDATION_TIME_LIMIT">15</setting>
    <setting id="AVAYA_DEBUG">true</setting>
    <setting id="AVAYA_DTMF_POSTFIX_TOKEN">#</setting>
    <setting id="AVAYA_DTMF_PREFIX_TOKEN">*95</setting>
    <setting id="AVAYA_TOKEN_LENGTH">6</setting>
    <setting id="AVAYA_MAX_DOWNLOAD_TRIES">3</setting>
    <setting id="AVAYA_DISABLE_AUDIO_RECORDING">false</setting>
    <dial-in-sequence>
      <conf-num>6000</conf-num>
      <delay>3000</delay>
      <dtmf>{x-tel-avaya-participant-code}</dtmf>
      <dtmf>#</dtmf>
      <delay>12000</delay>
      <dtmf>[uv-token]</dtmf>
      <dtmf>#</dtmf>
    </dial-in-sequence>
  </telephony-adaptor>

Basic Avaya adaptor settings

Adapter Setting Required Description
Avaya Adaptor AVAYA_BRIDGE_NAME Yes The name or IP address of the Avaya bridge server.
Avaya Adaptor AVAYA_PHONEOPERATOR_ID Yes The operator number of the operator channel to associate with the current session between Connect Pro and Avaya bridge. When a value of 0 is used, the bridge associates the next available operator channel with this session. If the value is less than 0 or greater than the highest defined operator number, the bridge associates the next available operator channel with this session.
Avaya Adaptor AVAYA_LOGINID Yes The user name used to establish a valid session with the bridge.
Avaya Adaptor AVAYA_PASSWORD Yes The password associated with the user name.
Avaya Adaptor AVAYA_FTP_DIRECTORY Yes The FTP directory for audio files on the Avaya bridge.
Avaya Adaptor AVAYA_FTP_LOGIN Yes The FTP user name.
Avaya Adaptor AVAYA_FTP_PASSWORD Yes The FTP password.
Avaya Adaptor AVAYA_AUDIO_CONVERTER_PATH Yes The path to the Avaya audio converter.

Advanced Avaya adaptor settings

Adapter Setting Required Default Value Description
Avaya Adaptor AVAYA_MAX_DOWNLOAD_ TRIES No 1440 The maximum number of times(attempts) Connect Pro tries to download audio recording file from Avaya telephony bridge. 
Avaya Adaptor AVAYA_DISABLE_AUDIO_ RECORDING No FALSE A Boolean value indicating whether audio recording is supported (FALSE) or not  (TRUE).   Using "TRUE" is helpful in cases when due to some reason audio recording is not working/supported on telephony bridge. If this setting is "TRUE" and adaptor gets a  start recording request, then the adaptor ignores this request. It shows a notification in meeting's user interface that audio recording is not supported on the telephony bridge.
Avaya Adaptor AVAYA_MUSICSOURCE No 0 The music source to play while participants are on hold. 
Avaya Adaptor AVAYA_VALIDATION_ TIME_LIMIT No 10 The maximum time, in seconds, to wait while validating conference. While creating/editing a telephony profile, adaptor validates the entered conference information from telephony bridge.  Adaptor waits for telephony bridge's response until validation time limit, after timeout Adaptor considers the profile validation as unsuccessful.  
Avaya Adaptor AVAYA_PHONE_PREFIX No null Prefix to the phone number. Use this setting if the system places “1” or “9” in front of a phone number. Adaptor adds this prefix in the telephone number (dialed from Connect meeting) before sending the  dial request to telephony bridge. 
Avaya Adaptor AVAYA_FILE_TRANSFER No autodetect Mode of file transfer (ftp or sftp) for audio recording download. Default value autodetect automatically decides the mode of file transfer(ftp/sftp) based on Avaya bridge version [ftp mode for Avaya bridge version 4.* and sftp mode is for Avaya bridge version >= 5.*].
Avaya Adaptor AVAYA_DTMF_PREFIX_ TOKEN No *95 Characters that indicate the DTMF entry is a token. Change this value only if the value has been changed on the bridge.
Avaya Adaptor AVAYA_TOKEN_LENGTH No 6 Number of digits in the unique token Connect generates for each user attending a meeting. 
Avaya Adaptor AVAYA_DTMF_POSTFIX_ TOKEN No # Characters that indicate the token is completed, which signals Connect  to generate the token to merge the phone user with a web user. Change this value only if the value has been changed on the bridge.
Avaya Adaptor AVAYA_DTMF_CAPTURE_ MAX_TIMEOUT No   Does not do anything.
Avaya Adaptor AVAYA_DTMF_CAPTURE_ MAX_LENGTH No   Does not do anything.
Avaya Adaptor AVAYA_DTMF_CAPTURE_ EXPIRE_ANNUNCIATOR_ NUM No   Does not do anything.

The following sample uses the dial-in sequence for the MeetingOne adaptor

<telephony-adaptor id="meetingone-adaptor" class-name="com.meetingone.adobeconnect.MeetingOneAdobeConnectAdaptor">
        <capabilities>
            <breeze-capabilities>
                <!-- "dial-out" and "dial-out-by-user" cannot both be enabled -->
                <!-- see also "call-selected-user" below -->
                <!-- if "dial-out" is true, then user's permission to dial out
                is determined by the user's role -->
                <capability id="dial-out" enabled="true">
                    <host enabled="true"/>
                    <presenter enabled="true"/>
                    <participant enabled="true"/>
                </capability>
                <!-- if "dial-out-by-user" is true, then user's permission
                to dial out is set for the individual user
                via xml api calls.  "dial-out" and "dial-out-by-user"
                are mutually exclusive capabilities -->
                <capability id="dial-out-by-user" enabled="false"/>

                <!-- if "auto-call-me-dialog" is true, then at start of
                meeting, the "Call Me" dialog box will pop up, if
                the user has permission to dial out -->
                <capability id="auto-call-me-dialog" enabled="true" />
                <!-- perform number masking using regular expression search and replacement. Default expressions mask digits 5,6,7 counting from last
                     ignoring any hyphens and spaces in between. The default expression match fails if there are less than 7 digits in the number. -->
                <capability id="number-mask" enabled="true">
                    <search-expression>([0-9])([- ]*)([0-9])([- ]*)([0-9])([- ]*)([0-9])([- ]*)([0-9])([- ]*)([0-9])([- ]*)([0-9])$</search-expression>
                    <replacement-expression>x$2x$4x$6$7$8$9$10$11$12$13</replacement-expression>
                </capability>
            </breeze-capabilities>
            <bridge-capabilities>
                <capability id="hang-up" enabled="true" />
                <capability id="remove-selected-user-enable-hangup" enabled="true" />
                <capability id="hold-user" enabled="true" />
                <capability id="volume-control" enabled="true"/>
                <capability id="mute-conference" enabled="true"/>
                <capability id="token-merge" enabled="true"/>
                <capability id="telephone-number-hint-format" value="E164"/>   

                <!-- audio+web breakout rooms. -->
                <capability id="web-audio-breakouts" enabled="true"/>

                <!-- true means meeting ui has no menu item to explicitly
                start audio conference.  Instead, conference start
                coincides automatically with meeting start -->
                <capability id="auto-start-conference" enabled="false"/>

                <!-- true means meeting ui has no menu item to explicitly
                stop audio conference.  Instead, conference stop
                coincides automatically with meeting end -->
                <capability id="auto-stop-conference" enabled="false"/>

                <!-- true means meeting ui allows call out to selected user -->
                <!-- see also "dial-out", "dial-out-by-user", "auto-call-me-dialog" above -->
                <capability id="call-selected-user" enabled="true"/>
            </bridge-capabilities>
        </capabilities>
    </telephony-adaptor>

Basic MeetingOne adaptor settings 

Adapter Setting Required Description
MeetingOne m1.connect.telephony.api_server Yes The URL of the MeetingOne telephony API server.
MeetingOne m1.connect.ftp.ssh Yes A Boolean value indicating whether SSH download is enabled (TRUE) or disabled (FALSE). The default value is TRUE.
MeetingOne m1.connect.loglevel Yes The logging level. Value can be info or debug  depending of level of debugging needed with debug level being the extreme l
MeetingOne m1.connect.telephony.api_



server.login
Yes The ID used to log in to the MeetingOne telephony API server.
MeetingOne m1.connect.telephony.api_



server.password
Yes  The password associated with the login ID.

Advanced MeetingOne adaptor settings

Adapter Setting Required Default Value Description
MeetingOne Adaptor MEETINGONE_DTMF_



PREFIX_TOKEN
Yes   Characters that indicate the DTMF entry is a token. Change this value only if the value has been changed on the bridge. Suggested value is *65*
MeetingOne Adaptor MEETINGONE_TOKEN_LENGTH Yes   Number of digits in the unique token Connect generates for each user attending a meeting. Suggested value is 4.
MeetingOne Adaptor MEETINGONE_DTMF_



POSTFIX_TOKEN
Yes   Characters that indicate the token is completed, which signals Connect  to generate the token to merge the phone user with a web user. Change this value only if the value has been changed on the bridge. Suggested value is #.
MeetingOne Adaptor m1.connect.ftp.delay No 57,600 (16 hours)  Maximum length of audio download file, in seconds. Set the minimum to 3600 (one hour)
MeetingOne Adaptor m1.connect.message.timeout No 30 Maximum time for command acknowledgment from audio bridge in seconds.  Recommended value is from 30 through 120.
MeetingOne Adaptor m1.connect.recording.enabled No TRUE  Boolean value that specifies whether recording is enabled.

 Additional information

XML element Description
<conf-num> The phone number for the audio conference. This element must be first in the dial-in-sequence. You can only have one <conf-num> element. The adaptor provides the value in curly brackets {}. 
<delay> A delay in the dialing sequence, in milliseconds.
<dtmf> A DTMF (dual-tone multi-frequency) tone. A DTMF value can be any number or letter on a telephony keypad, including * and #. 

 

 

 

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