Configure Adobe Connect telephony adaptors

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:

  1. In a text editor, open [root_install_dir]\TelephonyService\conf\telephony-settings.xml file.
  2. In the XML file, define the dial-in sequence to the audio conference provider.
  3. Validate and save the XML file.
  4. 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>

LoopUp adaptor settings

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>

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