Learn how to do advanced configurations to telephony adaptors along with their settings and samples.
When you install the Adobe Connect, all telephony adaptors are automatically installed, irrespective of whether you select them in the installer or not. If you select the adaptor in the installer, the configuration screen for that adaptor is displayed. You can configure the adaptor at the time of installation. The installer only performs the basic configuration for the telephony adaptors. You can perform the advanced configurations manually.
For more information, see Install and Configure Adobe Connect.
Configure telephony adaptors
To configure the telephony adaptors, after you have run the installer, do the following:
- In a text editor, open [root_install_dir]\TelephonyService\conf\telephony-settings.xml file.
- In the XML file, define the dial-in sequence to the audio conference provider.
- Validate and save the XML file.
- Restart Adobe Connect Central Application Server.
In the sample settings, the values provided in curly brackets ({}) are placeholders. When updating the settings, you need to provide actual values for these parameters.
Arkadin adaptor settings
The settings for Arkadin adaptor are provided below.
Setting |
Required |
Default Value |
Description |
ARKADIN_TOKEN_LENGTH |
Yes |
4 |
Length of the token for user identification and merging in the meeting room |
ARKADIN_DTMF_PREFIX_TOKEN |
Yes |
99 |
Prefix of the token for user identification and merging in the meeting room |
ARKADIN_UVLINE_CLIS |
Yes |
NA |
List of UV numbers/SIP account numbers used to join audio conference. |
ARKADIN_DTMF_POSTFIX_TOKEN |
Yes |
# |
Postfix of the token for user identification and merging in the meeting room |
ARKADIN_APPID |
Yes |
NA |
Arkadin provides this application ID |
ARKADIN_LOADBALANCER |
Yes |
NA |
Load balancer URL of the Arkadin bamboo server |
ARKADIN_RESPONSE_URL |
Yes |
NA |
Call back URL on which Arkadin can make callback to Connect Telephony Service |
ARKADIN_ACCESS_NUMBER_URL |
Yes |
NA |
The URL for the page displaying more dial- in numbers |
ARKADIN_AUTHENTICATE_URL |
Yes |
NA |
Authentication URL |
ARKADIN_BAMBOO_TIMEOUT |
Yes |
360000 |
Timeout for the Arkadin bamboo service |
MAX_SUB_CONFS |
Yes |
8 |
Maximum number of breakouts supported |
MAX_USERS_PER_SUB_CONF |
Yes |
100 |
Maximum number of users allowed per breakout |
ARKADIN_TOLLFREE |
Yes |
NA |
Set the attribute to true to display the tollfree number and false to not display the number. |
Arkadin adaptor sample settings for NA region
The following example demonstrates the dial-in sequence for Basic Arkadin adaptor. The value of the ARKADIN_TOLLFREE is different for NA region.
<telephony-settings>
<telephony-adaptor id="arkadin-adaptor" class- name="com.macromedia.breeze_ext.arkadin.ArkadinAdaptor" enabled="true">
<setting id="ARKADIN_UVLINE_CLIS">14158322000,4158322000,4085366000,4085366001,4155130607,4156589626,14156589626</setting>
<setting id="ARKADIN_TOKEN_LENGTH">4</setting>
<setting id="ARKADIN_DTMF_PREFIX_TOKEN">99</setting>
<setting id="ARKADIN_DTMF_POSTFIX_TOKEN">#</setting>
<setting id="ARKADIN_APPID">${Arkadin_APP_ID}</setting>
<setting id="ARKADIN_LOADBALANCER">${Arkadin_Bamboo_Server_Loadbalancer}</setting>
<setting id="ARKADIN_RESPONSE_URL">${Arkadin_Client_Callback_URL}</setting>
<setting id="ARKADIN_ACCESS_NUMBER_URL">${Arkadin_More_DialIn_Info_URL}</setting>
<setting id="ARKADIN_AUTHENTICATE_URL">${Arkadin_Auth_Host}</setting>
<setting id="ARKADIN_BAMBOO_TIMEOUT">360000</setting>
<setting id="MAX_SUB_CONFS">20</setting>
<setting id="ARKADIN_TOLLFREE">true</setting>
<setting id="MAX_USERS_PER_SUB_CONF">100</setting>
<dial-in-sequence>
<conf-num>{x-tel-arkadin-conference-number-free}</conf-num>
<delay>2000</delay>
<dtmf>{x-tel-arkadin-moderator-code}</dtmf>
<dtmf>#</dtmf>
<delay>500</delay>
<dtmf>#</dtmf>
<delay>3000</delay>
</dial-in-sequence>
</telephony-adaptor>
</telephony-settings>
Arkadin adaptor sample settings for APAC region
The following example demonstrates the dial-in sequence for Basic Arkadin adaptor. The dial sequence is different for APAC region.
<telephony-settings>
<telephony-adaptor id="arkadin-adaptor" class- name="com.macromedia.breeze_ext.arkadin.ArkadinAdaptor" enabled="true">
<setting id="ARKADIN_UVLINE_CLIS">14158322000,4158322000,4085366000,4085366001,4155130607,4156589626,14156589626</setting>
<setting id="ARKADIN_TOKEN_LENGTH">4</setting>
<setting id="ARKADIN_DTMF_PREFIX_TOKEN">99</setting>
<setting id="ARKADIN_DTMF_POSTFIX_TOKEN">#</setting>
<setting id="ARKADIN_APPID">${Arkadin_APP_ID}</setting>
<setting id="ARKADIN_LOADBALANCER">${Arkadin_Bamboo_Server_Loadbalancer}</setting>
<setting id="ARKADIN_RESPONSE_URL">${Arkadin_Client_Callback_URL}</setting>
<setting id="ARKADIN_ACCESS_NUMBER_URL">${Arkadin_More_DialIn_Info_URL}</setting>
<setting id="ARKADIN_AUTHENTICATE_URL">${Arkadin_Auth_Host}</setting>
<setting id="ARKADIN_BAMBOO_TIMEOUT">360000</setting>
<setting id="MAX_SUB_CONFS">20</setting>
<setting id="MAX_USERS_PER_SUB_CONF">100</setting>
<dial-in-sequence>
<conf-num>{x-tel-arkadin-conference-number-uvline}</conf-num>
<delay>2000</delay>
<dtmf>{x-tel-arkadin-moderator-code}</dtmf>
<dtmf>#</dtmf>
<delay>5000</delay>
<dtmf>6*</dtmf>
<dtmf>#</dtmf>
<delay>5000</delay>
<dtmf>6*</dtmf>
<dtmf>#</dtmf>
</dial-in-sequence>
</telephony-adaptor>
</telephony-settings>
PGi adaptor settings
The settings for Basic and Advanced PGi adaptors are described below.
Basic PGi adaptor settings for North America
Adapter |
Setting |
Required |
Default value |
Description |
Pgi NA |
PREMIERE_UV_COUNTRY_CODE |
No |
^USA.* |
Regular expression for matching the country code whose conference number is used for Universal Voice (UV). |
Advanced PGi adaptor settings for North America
Setting |
Required |
Default value |
Description |
PREMIERE_UV_NUMBER_TYPE |
No |
TOLL_FREE |
The number type of the conference number which is used for UV. |
PREMIERE_DISPLAY_COUNTRY_CODE |
No |
USAUSA, CANMON, CANTOR |
Comma-separated list of country code for which the conference numbers is stored and displayed in Adobe Connect. |
PREMIERE_DISPLAY_NUMBER_TYPES |
No |
TOLL, TOLL_FREE, LOCAL |
Comma-separated list of number types for which the conference numbers is stored and displayed in Connect. The possible number types are: • INTERNATIONAL_TOLL_FREE • LOCAL • LO_CALL • NATIONAL_FREE_PHONE • TOLL • TOLL_FREE |
PREMIERE_MAX_NUMBERS |
No |
20 |
Maximum number of phone numbers allowed to be stored in Adobe Connect. |
PGi adaptor for North America sample settings
The following example demonstrates the dial-in sequence for PGi adaptor for NA.
<telephony-settings>
<telephony-adaptor id="premiere-adaptor" class- name="com.macromedia.breeze_ext.premiere.gateway.PTekGateway" enabled="true" name="{premiere- adaptor}" default-provider="true">
<setting id="TOKEN_LENGTH">4</setting>
<setting id="MAX_SUB_CONFS">9</setting>
<setting id="MAX_USERS_PER_SUB_CONF">200</setting>
<setting id="DTMF_PREFIX_TOKEN">*29</setting>
<setting id="DTMF_POSTFIX_TOKEN">#*</setting>
<!--The replacement for '+' symbol in a dial-out number for simulating E164 support-->
<setting id="PREMIERE_PLUS_REPLACEMENT_FOR_E164">011</setting>
<!--Host name for the Premiere bridge. E.g. csrsaxis.premconf.com-->
<setting id="PREMIERE_HOST">${PREMIERE_HOST}</setting>
<!--Port number for the Premiere bridge. E.g. 443-->
<setting id="PREMIERE_PORT">${PREMIERE_PORT}</setting>
<!--Id for acessing Premiere bridge from adaptor-->
<setting id="PREMIERE_WEB_ID">${PREMIERE_WEB_ID}</setting>
<!--Password for acessing Premiere bridge from adaptor-->
<setting id="PREMIERE_PASSWORD">${PREMIERE_PASSWORD}</setting>
<!--Url for downloading recording from Premiere bridge from adaptor-->
<setting id="PREMIERE_DOWNLOAD_URL">${PREMIERE_DOWNLOAD_URL}</setting>
<!--Id for downloading recording from Premiere bridge from adaptor-->
<setting id="PREMIERE_DOWNLOAD_LOGIN">${PREMIERE_DOWNLOAD_LOGIN}</setting>
<!--Password for downloading recording from Premiere bridge from adaptor-->
<setting id="PREMIERE_DOWNLOAD_PASSWORD">${PREMIERE_DOWNLOAD_PASSWORD}</setting>
<!--Maximum number of allowed recording download attempts from Premiere bridge in case of failure.
E.g. 1440-->
<setting id="PREMIERE_MAX_DOWNLOAD_TRIES">${PREMIERE_MAX_DOWNLOAD_TRIES}</setting>
<!--Interval in milliseconds at which the scheduler sends status signal to PGI. Default is 60000.-->
<!--<setting id="PREMIERE_HEALTHCHECK_INTERVAL">${PREMIERE_HEALTHCHECK_INTERVAL}</setting>-->
<!--Interval in milliseconds at which the scheduler prints a report. Default is 600000.
E.g. 10000-->
<setting id="PREMIERE_REPORT_INTERVAL">${PREMIERE_REPORT_INTERVAL}</setting>
<!--Interval in milliseconds at which the scheduler attempts to download recording.
Default is 60000-->
<!-- <setting id="PREMIERE_RECORDING_DOWNLOAD_INTERVAL">${PREMIERE_RECORDING_DOWNLOAD_INTERVAL}</setting>
-->
<!--Maximum number of allowed attempts to connect to Premiere bridge in case of failure. Default is 3-->
<!--<setting id="PREMIERE_MAX_CONNECTION_TRIES">${PREMIERE_MAX_CONNECTION_TRIES}</setting>-->
<!--Interval in milliseconds for which adaptor waits for a response from Premiere bridge. E.g. 20000-->
<setting id="PREMIERE_TELEPHONY_TIMEOUT">${PREMIERE_TELEPHONY_TIMEOUT}</setting>
<!--The country code whose conference number is to be used for UV. Default is ^USA.* For EMEA geo region, you may override it to ^GBR.*-->
<setting id="PREMIERE_UV_COUNTRY_CODE">${PREMIERE_UV_COUNTRY_CODE}</setting>
<!--The number type of the conference number which should to be used for UV. Default is TOLL_FREE. For EMEA geo region, you may override it to INTERNATIONAL_TOLL_FREE-->
<!--<setting id="PREMIERE_UV_NUMBER_TYPE">${PREMIERE_UV_NUMBER_TYPE}</setting>-->
<!-- Comma seperated list of country code for which the conference numbers should be stored and displayed in Connect.
Default is USAUSA,CANMON,CANTOR. For EMEA geo region, you may override it to BELBSL,DNKCPN,FRAPRS,DEUFRK,INDDEL,IRLDUB,ITAROM,NLDASD,NOROSL,ESPMRD,SWESKM,CHEGNV,GBRLON,U SAUSA-->
<!--<setting id="PREMIERE_DISPLAY_COUNTRY_CODES">{PREMIERE_DISPLAY_COUNTRY_CODES}</setting>-->
<!-- Comma seperated list of number types for which the conference numbers should be stored and displayed in Connect.
The possible number types are: INTERNATIONAL_TOLL_FREE,LOCAL,LO_CALL,NATIONAL_FREE_PHONE,TOLL,TOLL_FREE. Default is
TOLL,TOLL_FREE. For EMEA, you may override it to LOCAL,INTERNATIONAL_TOLL_FREE,NATIONAL_FREE_PHONE,LO_CALL-->
<!--<setting id="PREMIERE_DISPLAY_NUMBER_TYPES">{PREMIERE_DISPLAY_NUMBER_TYPES}</setting>-->
<!--This is more info parameterized url. for now following parameters can be used x-tel-premiere-moderator-code for moderator code
x-tel-premiere-user-id premiere id
x-tel-premiere-participant-code participant code
x-tel-premiere-conference-number conference number
There is one example below with participant code and conference number -->
<!--setting id="PREMIRE_MORE_INFO_URL">https://www.myrcplus.com/cnums.asp?bwebid=8369444&ppc={x-tel- premiere-participant-code}&num={x-tel-premiere-conference-number}</setting-->
<setting id="PREMIERE_MORE_INFO_URL">https://www.myrcplus.com/cnums.asp?bwebid=8369444&ppc={x- tel-premiere-participant-code}&num={x-tel-premiere-conference-number}</setting>
<!--Maximum number of fields allowed to store that are created to store the list of conference numbers. Default is 6. Relevant only for EMEA-->
<!--<setting id="PREMIERE_MAX_NUMBER_FIELDS">${PREMIERE_MAX_NUMBER_FIELDS}</setting>-->
<!--Maximum number of phone numbers allowed to be stored in Connect. Default is 20.-->
<!--<setting id="PREMIERE_MAX_NUMBERS">${PREMIERE_MAX_NUMBERS}</setting>-->
<!--Flag to indiacate if this is EMEA. Set to yes/no. Default is 'no'-->
<!--<setting id="PREMIERE_EMEA">${PREMIERE_EMEA}</setting>-->
<!--Marker to identify EMEA conference nnumbers. Default is '_'.Relevant only for EMEA-
->
<!--<setting
id="PREMIERE_EMEA_NUMBER_MARKER">${PREMIERE_EMEA_NUMBER_MARKER}</setting>-->
<!--Flag to indiacate wether to use fallback algorithm for storing conference numbers. Set to yes/no. Default is 'no'-->
<!--<setting id="PREMIERE_EMEA_NUMBER_ALG_FALLBACK">${PREMIERE_EMEA_NUMBER_ALG_FALLBACK}</setting>-->
<!--Flag for enabling/disabling debugging for this adaptor.Set to yes/no. Default is 'yes'-->
<!--<setting id="PREMIERE_DEBUG_ALL">${DISABLE_DIALOUT}</setting>-->
1000.
<!--Time interval in millisecods for sending talker messages to Connect.Default is
Set to 0 or -1 to disable throttling.-->
<!--<setting
id="PREMIERE_TALKER_MESSAGE_THROTTLE_THRESHOLD">${PREMIERE_TALKER_MESSAGE_THROTTLE_THRESHOLD
}</setting>-->
<dial-in-sequence>
<conf-num>{x-tel-premiere-uv-conference-number}</conf-num>
<delay>6000</delay>
<dtmf>{x-tel-premiere-participant-code}</dtmf>
<dtmf>#</dtmf>
<delay>12000</delay>
<dtmf>*#</dtmf>
<delay>12000</delay>
<dtmf>[uv-token]</dtmf>
</dial-in-sequence>
</telephony-adaptor>
</telephony-settings>
Basic PGi adaptor settings for EMEA
Adapter |
Setting |
Required |
Default value |
Description |
Pgi EMEA |
EMEA_PREMIERE_UV_COUNTRY_CODE | No |
^GBR.* |
This Regular expression matches the conference number of the country code used for UV. |
Advanced PGi adaptor settings for EMEA
Adapter |
Setting |
Required |
Default value |
Description |
Pgi EMEA |
EMEA_PREMIERE_UV_NUMBER_TYPE | No |
NATIONAL_FREE_PHONE |
The number type of the conference number which is used for UV. |
Pgi EMEA |
EMEA_PREMIERE_MAX_NUMBERS |
No |
20 |
Maximum number of phone numbers allowed to be stored in Adobe Connect. |
PGi adaptor for EMEA sample settings
The following example demonstrates the dial-in sequence for PGi adaptor for EMEA.
<telephony-settings>
<telephony-adaptor id="premiere-emea-adaptor" class- name="com.macromedia.breeze_ext.premiere.gateway.EMEA.PTekGateway" enabled="true" default- provider="true">
<setting id="TOKEN_LENGTH">4</setting>
<setting id="MAX_SUB_CONFS">9</setting>
<setting id="MAX_USERS_PER_SUB_CONF">200</setting>
<setting id="DTMF_PREFIX_TOKEN">*29</setting>
<setting id="DTMF_POSTFIX_TOKEN">#*</setting>
<!--The replacement for '+' symbol in a dial-out number for simulating E164 support-->
<setting id="EMEA_PREMIERE_PLUS_REPLACEMENT_FOR_E164">011</setting>
<!--Host name for the Premiere bridge. E.g. csrsaxis.premconf.com-->
<setting id="EMEA_PREMIERE_HOST">${EMEA_PREMIERE_HOST}</setting>
<!--Port number for the Premiere bridge. E.g. 443-->
<setting id="EMEA_PREMIERE_PORT">${EMEA_PREMIERE_PORT}</setting>
<!--Id for acessing Premiere bridge from adaptor-->
<setting id="EMEA_PREMIERE_WEB_ID">${EMEA_PREMIERE_WEB_ID}</setting>
<!--Password for acessing Premiere bridge from adaptor-->
<setting id="EMEA_PREMIERE_PASSWORD">${EMEA_PREMIERE_PASSWORD}</setting>
<!--Url for downloading recording from Premiere bridge from adaptor-->
<setting id="EMEA_PREMIERE_DOWNLOAD_URL">${EMEA_PREMIERE_DOWNLOAD_URL}</setting>
<!--Id for downloading recording from Premiere bridge from adaptor-->
<setting id="EMEA_PREMIERE_DOWNLOAD_LOGIN">${EMEA_PREMIERE_DOWNLOAD_LOGIN}</setting>
<!--Password for downloading recording from Premiere bridge from adaptor-->
<setting id="EMEA_PREMIERE_DOWNLOAD_PASSWORD">${EMEA_PREMIERE_DOWNLOAD_PASSWORD}</setting>
<!--Maximum number of allowed recording download attempts from Premiere bridge in case of failure.
E.g. 1440-->
<setting id="EMEA_PREMIERE_MAX_DOWNLOAD_TRIES">${EMEA_PREMIERE_MAX_DOWNLOAD_TRIES}</setting>
<!--Interval in milliseconds at which the scheduler sends status signal to PGI. Default is 60000.-->
<!--<setting id="EMEA_PREMIERE_HEALTHCHECK_INTERVAL">${EMEA_PREMIERE_HEALTHCHECK_INTERVAL}</setting> -->
<!--Interval in milliseconds at which the scheduler prints a report. Default is 600000.
E.g. 10000-->
<setting id="EMEA_PREMIERE_REPORT_INTERVAL">${EMEA_PREMIERE_REPORT_INTERVAL}</setting>
<!--Interval in milliseconds at which the scheduler attempts to download recording.
Default is 60000. -->
<!-- <setting id="EMEA_PREMIERE_RECORDING_DOWNLOAD_INTERVAL">${EMEA_PREMIERE_RECORDING_DOWNLOAD_INTERVAL}<
/setting> -->
<!--Maximum number of allowed attempts to connect to Premiere bridge in case of failure. Default is 3-->
<!--<setting id="EMEA_PREMIERE_MAX_CONNECTION_TRIES">${EMEA_PREMIERE_MAX_CONNECTION_TRIES}</setting>-->
<!--Interval in milliseconds for which adaptor waits for a response from Premiere bridge. E.g. 20000-->
<setting id="EMEA_PREMIERE_TELEPHONY_TIMEOUT">${EMEA_PREMIERE_TELEPHONY_TIMEOUT}</setting>
<!--Country code whose conference number is to be used for UV. Default is ^GBR.* For NA geo region, you may override it to ^USA.*-->
<setting id="EMEA_PREMIERE_UV_COUNTRY_CODE">${EMEA_PREMIERE_UV_COUNTRY_CODE}</setting>
<!--The number type of the conference number which should to be used for UV. Default is NATIONAL_FREE_PHONE. For NA geo region, you may override it to TOLL_FREE-->
<!--<setting id="EMEA_PREMIERE_UV_NUMBER_TYPE">${EMEA_PREMIERE_UV_NUMBER_TYPE}</setting>-->
<!--This is more info parameterized url. for now following parameters can be used x-tel-premiere-moderator-code for moderator code
x-tel-premiere-user-id premiere id
x-tel-premiere-participant-code participant code
x-tel-premiere-conference-number conference number
There is one example below with participant code and conference number -->
<!--setting id="EMEA_PREMIRE_MORE_INFO_URL">https://www.myrcplus.com/cnums.asp?bwebid=8369444&ppc={x
-tel-premiere-participant-code}&num={x-tel-premiere-conference-number}</setting-->
<setting id="EMEA_PREMIERE_MORE_INFO_URL">http://www.eurcplus.com/cnums.asp?bwebid=8369444&ppc={x
-tel-premiere-participant-code}&num={x-tel-premiere-conference-number}</setting>
<!--Maximum number of fields allowed to store that are created to store the list of conference numbers. Default is 6. Relevant only for EMEA-->
<!--<setting id="EMEA_PREMIERE_MAX_NUMBER_FIELDS">${EMEA_PREMIERE_MAX_NUMBER_FIELDS}</setting>-->
<!--Maximum number of phone numbers allowed to be stored in Connect. Default is 20.-->
<!--<setting id="EMEA_PREMIERE_MAX_NUMBERS">${EMEA_PREMIERE_MAX_NUMBERS}</setting>--
>
<!--Flag to indiacate if this is EMEA. Set to yes/no. Default is 'no'-->
<!--<setting id="EMEA_PREMIERE_EMEA">${EMEA_PREMIERE_EMEA}</setting>-->
<!--Marker to identify EMEA conference nnumbers. Default is '_'.Relevant only for EMEA-
->
<!--<setting
id="EMEA_PREMIERE_EMEA_NUMBER_MARKER">${EMEA_PREMIERE_EMEA_NUMBER_MARKER}</setting>-->
<!--Flag to indiacate wether to use fallback algorithm for storing conference numbers. Set to yes/no. Default is 'no'-->
<!--<setting id="EMEA_PREMIERE_EMEA_NUMBER_ALG_FALLBACK">${EMEA_PREMIERE_EMEA_NUMBER_ALG_FALLBACK}</setti
ng>-->
<!--Flag for enabling/disabling debugging for this adaptor.Set to yes/no. Default is 'yes'-->
<!--<setting id="EMEA_PREMIERE_DEBUG_ALL">${DISABLE_DIALOUT}</setting>-->
1000.
<!--Time interval in millisecods for sending talker messages to Connect.Default is
Set to 0 or -1 to disable throttling.-->
<!--<setting
id="EMEA_PREMIERE_TALKER_MESSAGE_THROTTLE_THRESHOLD">${EMEA_PREMIERE_TALKER_MESSAGE_THROTTLE
_THRESHOLD}</setting>-->
<dial-in-sequence>
<conf-num>{x-tel-premiere-emea-uv-conference-number}</conf-num>
<delay>6000</delay>
<dtmf>{x-tel-premiere-emea-participant-code}</dtmf>
<dtmf>#</dtmf>
<delay>12000</delay>
<dtmf>*#</dtmf>
<delay>12000</delay>
<dtmf>[uv-token]</dtmf>
</dial-in-sequence>
</telephony-adaptor>
</telephony-settings>
Intercall adaptor settings
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 is 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 |
15,000 (15 seconds) |
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 time to skip before checking for the activeness of a meeting in Adobe Connect. This ensures that sessions do not continue to linger forever. Also the sessions 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 that the token is completed. This action signals Adobe 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. |
InterCall Adaptor sample settings
The following example demonstrates the dial-in sequence for the InterCall adaptor.
<telephony-settings>
<telephony-adaptor id="intercall-adaptor" class- name="com.macromedia.breeze_ext.telephony.Intercall.IntercallTelephonyAdaptor" enabled="true" disable-profiles-on-edit="true" disable-profiles-on-disable="true">
<setting id="TOKEN_LENGTH">4</setting>
<setting id="MAX_SUB_CONFS">15</setting>
<setting id="MAX_USERS_PER_SUB_CONF">200</setting>
<setting id="DTMF_PREFIX_TOKEN">#1</setting>
<setting id="DTMF_POSTFIX_TOKEN">#</setting>
<setting id="CONFERENCE_START_WAIT_TIME">20000</setting>
<setting id="INTERCALL_DEBUG">${INTERCALL_DEBUG}</setting>
<setting id="INTERCALL_HEARTBEAT_INTERVAL">${INTERCALL_HEARTBEAT_INTERVAL}</setting>
<setting id="INTERCALL_CCAPI_HOST">${INTERCALL_CCAPI_HOST}</setting>
<setting id="INTERCALL_CCAPI_AUTH_HOST">${INTERCALL_CCAPI_AUTH_HOST}</setting>
<setting id="INTERCALL_CLIENT_CALLBACK_URL">${INTERCALL_CLIENT_CALLBACK_URL}</setting>
<setting id="INTERCALL_APP_TOKEN">${INTERCALL_APP_TOKEN}</setting>
<setting id="INTERCALL_EMEA_COUNTRY_CODES">${INTERCALL_EMEA_COUNTRY_CODES}</setting>
<setting id="INTERCALL_TOLL_FREE_COUNTRY_CODE">${INTERCALL_TOLL_FREE_COUNTRY_CODE}</setting>
<dial-in-sequence>
<conf-num>{x-tel-intercall-uv-conference-number}</conf-num>
<delay>6000</delay>
<dtmf>{x-tel-intercall-participant-code}</dtmf>
<dtmf>#</dtmf>
<delay>8000</delay>
<dtmf>#</dtmf>
<delay>8000</delay>
<dtmf>#</dtmf>
<delay>12000</delay>
<dtmf>[uv-token]</dtmf>
</dial-in-sequence>
</telephony-adaptor>
</telephony-settings>
Avaya adaptor settings
The Avaya adapter is not maintained by Adobe Connect. The adaptor is shipped ‘as-is’ as received from Avaya. For any issues with installing the Avaya adapter, customers would have to reach out to the Avaya customer support for troubleshooting and software updates. A current maintenance contract with Avaya covering the audio bridge is required.
Basic Avaya adaptor settings
Adaptor |
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 Adobe Connect and Avaya bridge. When a value of 0 is used, the bridge associates the next available operator channel with this session. If the value used is either less than 0 or greater than the highest operator number defined on the bridge, 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
Adaptor |
Setting |
Required |
Default Value |
Description |
Avaya Adaptor |
AVAYA_MAX_DOWNLOAD_TRIES |
No |
1440 |
The maximum number of times (attempts) Adobe Connect tries to download an audio recording file from the Avaya telephony bridge. |
Avaya Adaptor |
AVAYA_DISABLE_AUDIO_RECORDING |
No |
False |
A Boolean value indicating whether audio recording is supported (FALSE) or not (TRUE). Setting this parameter to True can be helpful in cases where audio recording is not working or is not supported on telephony bridge. If this setting is True and the adaptor gets a start recording request, then the adaptor simply ignores this request. The adaptor also shows a notification in the 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 a conference. While creating/editing a telephony profile, the adaptor validates the entered conference information from the telephony bridge. The adaptor waits for the telephony bridge's response until validation time limit. After time-out, the 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. The adaptor adds this prefix in the telephone number (dialed from an Adobe Connect Meeting) before sending the dial request to the telephony bridge. |
Avaya Adaptor |
AVAYA_FILE_TRANSFER |
No |
autodetect |
Mode of file transfer (ftp or sftp) for audio recording download. The default value - autodetect - automatically decides the mode of file transfer (ftp/sftp) based on the 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. |
Avaya adaptor sample settings
The following example demonstrates the dial-in sequence for the Avaya adaptor.
<telephony-settings>
<telephony-adaptor id="avaya-adaptor" class- name="com.macromedia.breeze_ext.telephony.AvayaAdaptor" enabled="true">
<setting id="AVAYA_AUDIO_CONVERTER_PATH">${app.root}/util/avaya/</setting>
<setting id="AVAYA_BRIDGE_NAME">${AVAYA_BRIDGE_NAME}</setting>
<setting id="AVAYA_DISABLE_AUDIO_RECORDING">${AVAYA_DISABLE_AUDIO_RECORDING}</setting>
<setting id="AVAYA_DTMF_CAPTURE_MAX_TIMEOUT">${AVAYA_DTMF_CAPTURE_MAX_TIMEOUT}</setting>
<setting id="AVAYA_DTMF_CAPTURE_MAX_LENGTH">${AVAYA_DTMF_CAPTURE_MAX_LENGTH}</setting>
<setting id="AVAYA_DTMF_POSTFIX_TOKEN">${AVAYA_DTMF_POSTFIX_TOKEN}</setting>
<setting id="AVAYA_DTMF_PREFIX_TOKEN">${AVAYA_DTMF_PREFIX_TOKEN}</setting>
<setting id="AVAYA_FTP_DIRECTORY">${AVAYA_FTP_DIRECTORY}/</setting>
<setting id="AVAYA_FILE_TRANSFER">${AVAYA_FILE_TRANSFER}</setting>
<setting id="AVAYA_MAX_DOWNLOAD_TRIES">${AVAYA_MAX_DOWNLOAD_TRIES}</setting>
<setting id="AVAYA_FTP_LOGIN">${AVAYA_FTP_LOGIN}</setting>
<setting id="AVAYA_FTP_PASSWORD">${AVAYA_FTP_PASSWORD}</setting>
<setting id="AVAYA_LOGINID">${AVAYA_LOGINID}</setting>
<setting id="AVAYA_PASSWORD">${AVAYA_PASSWORD}</setting>
<setting id="AVAYA_MUSICSOURCE">${AVAYA_MUSICSOURCE}</setting>
<setting id="AVAYA_PHONEOPERATOR_ID">${AVAYA_PHONEOPERATOR_ID}</setting>
<setting id="AVAYA_TOKEN_LENGTH">${AVAYA_TOKEN_LENGTH}</setting>
<setting id="MAX_SUB_CONFS">9</setting>
<setting id="MAX_USERS_PER_SUB_CONF">200</setting>
<setting id="AVAYA_VALIDATION_TIME_LIMIT">${AVAYA_VALIDATION_TIME_LIMIT}</setting>
<dial-in-sequence>
<conf-num>${AVAYA_DNIS_NUMBER}</conf-num>
<delay>3000</delay>
<dtmf>{x-tel-avaya-participant-code}</dtmf>
<dtmf>#</dtmf>
</dial-in-sequence>
</telephony-adaptor>
</telephony-settings>
Additional information
Some common XML elements are described below.
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 the 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 telephone keypad, including * and #. |
MeetingOne adaptor settings
Basic settings
Setting |
Required |
Description |
m1.connect.telephony.api_server |
Yes |
The URL of the MeetingOne telephony API server. |
m1.connect.ftp.ssh |
Yes |
A Boolean value indicating whether SSH download is enabled (TRUE) or disabled (FALSE). The default value is TRUE. |
m1.connect.loglevel |
Yes |
The logging level. Value can be info or debug depending on the level of debugging needed with debug level being the extreme. |
m1.connect.telephony.api_ server.login |
Yes |
The ID used to log in to the MeetingOne telephony API server. |
m1.connect.telephony.api_ server.password |
Yes |
The password associated with the login ID. |
Advanced settings
Setting |
Required |
Default Value |
Description |
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_TOKEN_LENGTH |
Yes |
- |
Number of digits in the unique token that Connect generates for each user attending a meeting. Suggested value is 4. |
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 #. |
m1.connect.ftp.delay |
No |
57,600 (16 hours) |
Maximum length of audio download file, in seconds. Minimum is 3600 (one hour). |
m1.connect.message.timeout |
No |
30 |
Maximum time for command acknowledgement from audio bridge in seconds. Recommended value is between 30 and 120 seconds. |
m1.connect.recording.enabled |
No |
TRUE |
Boolean value that specifies whether recording is enabled. |
m1.connect.sshdownload.cmd |
yes |
${MEETINGONE_PSFTP_PATH} {0} {1} {2} {3} |
SSH Download Cmd |
m1.connect.telephony.authentication_service_endpoint |
no |
authentication |
Telephony Authentication Service Endpoint |
m1.connect.telephony.audio_service_endpoint |
no |
audio |
Telephony Audio Service Endpoint |
m1.connect.telephony.events_service_endpoint |
no |
events |
Telephony Audio Service Endpoint |
MAX_SUB_CONFS |
yes |
20 |
|
MAX_USERS_PER_SUB_CONF |
yes |
150 |
|
Sample settings
The following example demonstrates the dial-in sequence for the MeetingOne adaptor:
<telephony-settings>
<telephony-adaptor id="meetingone-adaptor" class- name="com.meetingone.adobeconnect.MeetingOneAdobeConnectAdaptor" enabled="true" name="{meetingone-adaptor}">
<setting id="MEETINGONE_TOKEN_LENGTH">${MEETINGONE_TOKEN_LENGTH}</setting>
<setting id="MAX_SUB_CONFS">20</setting>
<setting id="MAX_USERS_PER_SUB_CONF">150</setting>
<setting id="MEETINGONE_DTMF_PREFIX_TOKEN">${MEETINGONE_DTMF_PREFIX_TOKEN}</setting>
<setting id="MEETINGONE_DTMF_POSTFIX_TOKEN">${MEETINGONE_DTMF_POSTFIX_TOKEN}</setting>
<setting id="m1.connect.telephony.api_server">https://ape-secure.poweredbyphoenix.net/api</setting>
<setting id="m1.connect.ftp.ssh">${m1.connect.ftp.ssh}</setting>
<setting id="m1.connect.loglevel">${m1.connect.loglevel}</setting>
<setting id="m1.connect.sshdownload.cmd">${MEETINGONE_PSFTP_PATH} {0} {1} {2}
{3}</setting>
<setting id="m1.connect.recording.enabled">${m1.connect.recording.enabled}</setting>
<setting id="m1.connect.telephony.api_server.password">${m1.connect.telephony.api_server.password}
</setting>
<setting id="m1.connect.telephony.api_server.login">${m1.connect.telephony.api_server.login}</setting>
<setting id="m1.connect.telephony.authentication_service_endpoint">authentication</setting>
<setting id="m1.connect.telephony.audio_service_endpoint">audio</setting>
<setting id="m1.connect.telephony.events_service_endpoint">events</setting>
<setting id="m1.connect.ftp.delay">${m1.connect.ftp.delay}</setting>
<setting id="m1.connect.message.timeout">${m1.connect.message.timeout}</setting>
<setting id="DIALIN_NUMBERS">MeetingOne_Access_Number_Argentina MeetingOne_Access_Number_Australia MeetingOne_Access_Number_Austria MeetingOne_Access_Number_Bahrain MeetingOne_Access_Number_Belgium MeetingOne_Access_Number_Brazil MeetingOne_Access_Number_Bulgaria MeetingOne_Access_Number_Canada MeetingOne_Access_Number_China MeetingOne_Access_Number_Cyprus MeetingOne_Access_Number_Czech MeetingOne_Access_Number_Denmark MeetingOne_Access_Number_El_Salvador MeetingOne_Access_Number_Estonia MeetingOne_Access_Number_Finland MeetingOne_Access_Number_France MeetingOne_Access_Number_Germany MeetingOne_Access_Number_Greece MeetingOne_Access_Number_Hungary MeetingOne_Access_Number_Ireland MeetingOne_Access_Number_Israel MeetingOne_Access_Number_Italy MeetingOne_Access_Number_Japan MeetingOne_Access_Number_Latvia MeetingOne_Access_Number_Lithuania MeetingOne_Access_Number_Luxembourg MeetingOne_Access_Number_Mexico MeetingOne_Access_Number_Netherlands MeetingOne_Access_Number_New_Zealand MeetingOne_Access_Number_Norway MeetingOne_Access_Number_Panama MeetingOne_Access_Number_Peru MeetingOne_Access_Number_Poland MeetingOne_Access_Number_Portugal MeetingOne_Access_Number_Romania MeetingOne_Access_Number_Singapore MeetingOne_Access_Number_Slovenia MeetingOne_Access_Number_South_Africa MeetingOne_Access_Number_Spain MeetingOne_Access_Number_Sweden MeetingOne_Access_Number_Switzerland MeetingOne_Access_Number_United_Kingdom MeetingOne_Access_Number_United_States</setting>
<dial-in-sequence>
<conf-num>18008320736</conf-num>
<delay>12000</delay>
<dtmf>*</dtmf>
<dtmf>{x-tel-meetingone-conference-id}</dtmf>
<dtmf>#</dtmf>
<delay>6000</delay>
<dtmf>[uv-token]</dtmf>
</dial-in-sequence>
</telephony-adaptor>
</telephony-settings>
Basic settings - EMEA
Setting |
Required |
Description |
m1.connect.telephony.api_server |
Yes |
The URL of the MeetingOne EMEA telephony API server. |
m1.connect.ftp.ssh |
Yes |
A Boolean value indicating whether SSH download is enabled (TRUE) or is disabled (FALSE). The default value is TRUE. |
m1.connect.loglevel |
Yes |
The logging level. Value can be info or debug depending on the level of debugging needed with debug level being the extreme. |
m1.connect.telephony.api_server.login |
Yes |
The ID used to log in to the MeetingOne EMEA telephony API server. |
m1.connect.telephony.api_server.password |
Yes |
The password associated with the login ID. |
Advanced settings - EMEA
Setting |
Required |
Default Value |
Description |
MEETINGONE_DTMF_PREFIX_TOKEN |
Yes |
- |
Characters that Indicate that the DTMF entry is a token. Change this value only if the value has been changed on the bridge. A suggested value is *65. |
MEETINGONE_TOKEN_ LENGTH |
Yes |
- |
Number of digits in the unique token that Adobe Connect generates for each user attending a meeting. The suggested value is 4 |
MEETINGONE_DTMF_POSTFIX_TOKEN |
Yes |
- |
Characters that indicate the token is completed, which signals Adobe 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 #. |
m1.connect.ftp.delay |
No |
57,600 (16 hours) | Maximum length of audio download file in seconds. The minimum value is 3600 (one hour). |
m1.connect.message.timeout |
No |
30 |
The maximum time for command acknowledgement from audio bridge in seconds. Recommended value is between 30 and 120 seconds. |
m1.connect.recording.enabled |
No |
TRUE |
A Boolean value that specifies if recording is enabled. |
Sample settings - EMEA
The following example demonstrates the dial-in sequence for the MeetingOne EMEA adaptor:
<telephony-adaptor class-name="com.meetingone.adobeconnect.emea.AdaptorWrapper" enabled="true" id="meetingone-emea-adaptor" name="{meetingone-emea-adaptor}" use-backup- provider="true" default-recording-source="adaptor" disable-profiles-on-edit="false" disable- profiles-on-disable="false"><setting id="MEETINGONE_TOKEN_LENGTH">${MEETINGONE_TOKEN_LENGTH}</setting>
<setting id="MAX_SUB_CONFS">20</setting>
<setting id="MAX_USERS_PER_SUB_CONF">150</setting>
<setting id="MEETINGONE_DTMF_PREFIX_TOKEN">${MEETINGONE_DTMF_PREFIX_TOKEN}</setting>
<setting id="MEETINGONE_DTMF_POSTFIX_TOKEN">${MEETINGONE_DTMF_POSTFIX_TOKEN}</setting>
<setting id="m1.connect.telephony.api_server">${m1.connect.telephony.api_server}</setting>
<setting id="m1.connect.ftp.ssh">${m1.connect.ftp.ssh}</setting>
<setting id="m1.connect.loglevel">${m1.connect.loglevel}</setting>
<setting id="m1.connect.sshdownload.cmd">${MEETINGONE_PSFTP_PATH} {0} {1} {2}
{3}</setting>
<setting id="m1.connect.recording.enabled">${m1.connect.recording.enabled}</setting>
<setting id="m1.connect.telephony.api_server.password">${m1.connect.telephony.api_server.password}</s etting>
<setting id="m1.connect.telephony.api_server.login">${m1.connect.telephony.api_server.login}</setting
>
<setting id="m1.connect.telephony.authentication_service_endpoint">authentication</setting>
<setting id="m1.connect.telephony.audio_service_endpoint">audio</setting>
<setting id="m1.connect.telephony.events_service_endpoint">events</setting>
<setting id="m1.connect.ftp.delay">${m1.connect.ftp.delay}</setting>
<setting id="m1.connect.message.timeout">${m1.connect.message.timeout}</setting>
<setting id="DIALIN_NUMBERS">MeetingOne_Access_Number_Argentina MeetingOne_Access_Number_Australia MeetingOne_Access_Number_Austria MeetingOne_Access_Number_Bahrain MeetingOne_Access_Number_Belgium MeetingOne_Access_Number_Brazil MeetingOne_Access_Number_Bulgaria MeetingOne_Access_Number_Canada MeetingOne_Access_Number_China MeetingOne_Access_Number_Cyprus MeetingOne_Access_Number_Czech MeetingOne_Access_Number_Denmark MeetingOne_Access_Number_El_Salvador MeetingOne_Access_Number_Estonia MeetingOne_Access_Number_Finland MeetingOne_Access_Number_France MeetingOne_Access_Number_Germany MeetingOne_Access_Number_Greece MeetingOne_Access_Number_Hungary MeetingOne_Access_Number_Ireland MeetingOne_Access_Number_Israel MeetingOne_Access_Number_Italy MeetingOne_Access_Number_Japan MeetingOne_Access_Number_Latvia MeetingOne_Access_Number_Lithuania MeetingOne_Access_Number_Luxembourg MeetingOne_Access_Number_Mexico MeetingOne_Access_Number_Netherlands MeetingOne_Access_Number_New_Zealand MeetingOne_Access_Number_Norway
MeetingOne_Access_Number_Panama MeetingOne_Access_Number_Peru MeetingOne_Access_Number_Poland MeetingOne_Access_Number_Portugal MeetingOne_Access_Number_Romania MeetingOne_Access_Number_Singapore MeetingOne_Access_Number_Slovenia MeetingOne_Access_Number_South_Africa MeetingOne_Access_Number_Spain MeetingOne_Access_Number_Sweden MeetingOne_Access_Number_Switzerland MeetingOne_Access_Number_United_Kingdom MeetingOne_Access_Number_United_States</setting>
<dial-in-sequence>
<conf-num>{conf-num}</conf-num>
<delay>12000</delay>
<dtmf>*</dtmf>
<dtmf>{x-tel-meetingone-conference-id}</dtmf>
<dtmf>#</dtmf>
<delay>12000</delay>
<dtmf>*</dtmf>
<dtmf>{x-tel-meetingone-host-pin}</dtmf>
<dtmf>#</dtmf>
<delay>12000</delay>
<dtmf>[uv-token]</dtmf>
</dial-in-sequence>
</telephony-adaptor>
Sample capabilities - EMEA
The following example demonstrates the telephony-capabilities.xml configuration for the MeetingOne EMEA adaptor:
<telephony-adaptor class-name="com.meetingone.adobeconnect.emea.AdaptorWrapper" enabled="true" id="meetingone-emea-adaptor"> <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 enabled="true" id="dial-out"> <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 enabled="false" id="dial-out-by-user"/> <!-- 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 enabled="true" id="auto-call-me-dialog"/> <!-- 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 enabled="false" id="number-mask"> <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 enabled="true" id="hang-up"/> <capability enabled="true" id="remove-selected-user-enable-hangup"/> <capability enabled="true" id="hold-user"/> <capability enabled="true" id="volume-control"/> <capability enabled="true" id="mute-conference"/> <capability enabled="true" id="token-merge"/> <capability id="telephone-number-hint-format" value="E164"/> <!-- audio+web breakout rooms. --> <capability enabled="true" id="web-audio-breakouts"/> <!-- true means meeting ui has no menu item to explicitly start audio conference. Instead, conference start coincides automatically with meeting start --> <capability enabled="false" id="auto-start-conference"/> <!-- true means meeting ui has no menu item to explicitly stop audio conference. Instead, conference stop coincides automatically with meeting end --> <capability enabled="false" id="auto-stop-conference"/> <!-- true means meeting ui allows call out to selected user --> <!-- see also "dial-out", "dial-out-by-user", "auto-call-me-dialog" above--> <capability enabled="true" id="call-selected-user"/> </bridge-capabilities> </capabilities> </telephony-adaptor>
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