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>
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>
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 |
90 |
Maximum time for command acknowledgement from audio bridge in seconds. Recommended value is between 90 and 120 seconds. |
m1.connect.recording.enabled |
No |
TRUE |
Boolean value that specifies whether recording is enabled. |
m1.connect.sshdownload.cmd |
yes |
${MEETINGONE_PSFTP_PATH} {0} {1} {2} {3} |
SSH Download Cmd |
m1.connect.telephony.authentication_service_endpoint |
no |
authentication |
Telephony Authentication Service Endpoint |
m1.connect.telephony.audio_service_endpoint |
no |
audio |
Telephony Audio Service Endpoint |
m1.connect.telephony.events_service_endpoint |
no |
events |
Telephony Audio Service Endpoint |
MAX_SUB_CONFS |
yes |
20 |
|
MAX_USERS_PER_SUB_CONF |
yes |
150 |
|
Sample settings MeetingOne
The following example demonstrates the dial-in sequence for the MeetingOne adaptor:
<telephony-settings>
<telephony-adaptor id="meetingone-adaptor" class- name="com.meetingone.adobeconnect.MeetingOneAdobeConnectAdaptor" enabled="true" name="{meetingone-adaptor}">
<setting id="MEETINGONE_TOKEN_LENGTH">${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>
Sample capabilities MeetingOne
The following example demonstrates the telephony-capabilities.xml configuration for the MeetingOne adaptor:
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"/> <capability enabled="true" id="breakout-room"/> <!-- 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>
Setting |
Required |
Description |
| DS | Yes | Reserved. Must be present. Note that we set the first release version to 1.1 for deployment tracking. |
| DSCountry Code | Yes | The Data Center Country Code must be one of the following: USA -> US Toll free number for Universal Voice GBR -> UK Toll free number for Universal Voice |
| LogAltairResponse | Yes | Do not change |
| LogExtraDebug | Yes |
Do not change |
| LogSuppressAllByContract | Yes |
For Licensed Intall add this entry and set to true. This will suppress extra logging on Loopup endpoint. |
| AltairURL | Yes |
Default/base altair address. |
| AthenaURL | Yes |
LoopUp API server. |
| AthenaVerifySecret | Yes |
Password for LoopUp API Server. |
| SeqUrl | Yes |
Address of the Seq instance. |
| SeqToken | Yes |
SEQ token for Adobe Connect Adapter |
| DTMF_PREFIX_TOKEN |
No | LoopUp IVR token is via ##5 followed by # digit |
| DTMF_PREFIX_TOKEN |
No | Return to conference (token ends with..) |
| TOKEN_LENGTH |
No | 3-digit IVR token. Please do not change |
| MAX_SUB_CONFS |
Yes |
Set to 0. NO breakout support. |
| UVMaskDialinCLI_CSV |
Yes |
sip:RECORDING is required and ensures that the LoopUp recording leg (if present) is suppressed. |
Sample settings - LoopUp
The following example demonstrates the telephony-settings.xml configuration for the LoopUp telephony adaptor:
<?xml version="1.0" encoding="UTF-8"?>
<telephony-settings>
<telephony-adaptor id="LoopUp"
class-name="com.loopup.ACAdapter"
enabled="true"
name="LoopUp"
default-provider="true"
disable-profiles-on-edit="false"
disable-profiles-on-disable="false">
<!-- Configuration: Refer to
https://github.com/loopup/ACAdapter/blob/main/README.md#configuration -->
<setting id="DS">1.1</setting>
<setting id="DCCountryCode">USA</setting>
<setting id="LogAltairResponse">false</setting>
<setting id="LogExtraDebug">false</setting>
<setting id="AltairUrl">altair.loopup.com</setting>
<setting id="AthenaUrl">data.loopup.com</setting>
<!-- WARNING! This value is transformed by Adobe Telephony Service and will not match this
source! Authorotative file
https://github.com/loopup/ACAdapter/blob/main/conf/telephony-settings.xml -->
<setting id="AthenaVerifySecret_PASSWORD"></setting>
<setting id="SeqUrl">seq-altair.loopup.com</setting>
<!-- WARNING! This value is transformed by Adobe Telephony Service and will not match this
source! Authorotative file
https://github.com/loopup/ACAdapter/blob/main/conf/telephony-settings.xml -->
<setting id="SeqToken_PASSWORD"></setting>
<setting id="DTMF_PREFIX_TOKEN">##5</setting>
<setting id="DTMF_POSTFIX_TOKEN">#</setting>
<setting id="TOKEN_LENGTH">3</setting>
<setting id="MAX_SUB_CONFS">0</setting>
<!-- This is a comma seperated list contains the ANI/CLI of
Adobe's Universal Voice dialin numbers into the LoopUp bridge
which unless otherwise listed here will show up as attendees. -->
<setting id="UVMaskDialinCLI_CSV">sip:RECORDING</setting>
</telephony-adaptor>
</telephony-settings>
Sample capabilities - LoopUp
The following example demonstrates the telephony-capabilities.xml configuration for the LoopUp telephpny adaptor:
<?xml version="1.0" encoding="ISO-8859-1" standalone="no"?>
<telephony-capabilities>
<telephony-adaptor id="LoopUp" class-name="com.loopup.ACAdapter" enabled="true">
<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="false">
<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="false" />
<capability id="token-merge" enabled="true" />
<capability id="mute-all" enabled="false" />
<!-- audio + web breakout rooms. -->
<capability id="breakout-room" enabled="false" />
<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="true" />
<!-- 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" />
<capability id="telephone-number-hint-format" value="E164" />
</bridge-capabilities>
</capabilities>
</telephony-adaptor>
</telephony-capabilities>