Archive for April, 2014

How to make Connect use a browser-based email client to send Meeting invitations

When invoking a browser-based email client to invite participants to a Connect Meeting from within a Connect Meeting you will see this error message unless you first make the browser-based email your default email service:

 

bb-email.fw

 

Sending a browser-based email invitation from within a Connect Meeting is possible if you first make the browser-based email option your default email program. As an example, you can use the instructions at the following links to make Gmail your default email program :

http://email.about.com/od/gmailtips/qt/et_default_prog.htm

http://www.pcdailytips.com/set-gmail-as-default-email-using-chrome-or-firefox/

Note that once you enable a browser-based email client and invoke it from within a Connect Meeting, the behavior will be different based on whether the host issuing the invitation is using the Connect Meeting addin or the Flash Player. In the addin it will look like this:

Using the Connect Meeting addin, invoke the invitation: Meeting> Manage Access & Entry > Invite Participants

bb-email-1.fw

bb-email-2.fw

See how the invitation is fully populated with important details:

bb-email-3.fw

Following the same procedure using the Flash Player instead of the addin (?launcher=false) also works, but with an abbreviated invitation message:

bb-email-4.fw

 

Connect 9.1.x on-premise server – “Send Invitations” checked by default

When creating a new meeting you are asked if you want to send out meeting invitations by email.

In Connect 9.1.x the option to send invitations is selected by default.

If you do not wish to send out invitations for your meetings you have to select “Do not send invitations” every time you create a new meeting.

sendInvitations

You can change this behavior to make  “Do not send invitations” the default when creating a new meeting.

To do so, edit the notify.xsl file which is located in \Connect\9.1.1\appserv\apps\meeting\  ( but please remember to take a backup copy of the file).

1. Open the notify.xsl in an xml-friendly editor such as notepad++

2. Find this section:

<table cellpadding=”0″ cellspacing=”0″>
<xsl:call-template name=”input”>
<xsl:with-param name=”title”   select=”‘send-invitations'”/>
<xsl:with-param name=”name”    select=”‘date-scheduled'”/>
<xsl:with-param name=”type”    select=”‘radio'”/>
<xsl:with-param name=”value”   select=”/results/common/date”/>
<xsl:with-param name=”checked” select=”true()”/>
</xsl:call-template>

<xsl:call-template name=”input”>
<xsl:with-param name=”title”   select=”‘no-invitations'”/>
<xsl:with-param name=”name”    select=”‘date-scheduled'”/>
<xsl:with-param name=”type”    select=”‘radio'”/>
<xsl:with-param name=”value”   select=”‘ignore'”/>
<xsl:with-param name=”checked” select=”false()”/>
</xsl:call-template>
</table>

3. Change “false” to “true” and “true” to “false” to swap the selection.

It should now look like this:

<table cellpadding=”0″ cellspacing=”0″>
<xsl:call-template name=”input”>
<xsl:with-param name=”title”   select=”‘send-invitations'”/>
<xsl:with-param name=”name”    select=”‘date-scheduled'”/>
<xsl:with-param name=”type”    select=”‘radio'”/>
<xsl:with-param name=”value”   select=”/results/common/date”/>
<xsl:with-param name=”checked” select=”false()”/>
</xsl:call-template>

<xsl:call-template name=”input”>
<xsl:with-param name=”title”   select=”‘no-invitations'”/>
<xsl:with-param name=”name”    select=”‘date-scheduled'”/>
<xsl:with-param name=”type”    select=”‘radio'”/>
<xsl:with-param name=”value”   select=”‘ignore'”/>
<xsl:with-param name=”checked” select=”true()”/>
</xsl:call-template>
</table>

 

4. Save the file and restart the services.

5. Check your changes by creating a new meeting. If you encounter any issues, restore the original file.

 

 

Specifications for MP4 Conversion for Connect Recordings

Here are the specifications for the MP4 conversion; they are similar to our FLV specifications albeit with better compression:

  • Resolution: 1024X768
  • Frames Per Second: 8 FPS
  • Video Bitrate: 1024kbps
  • Audio:
    • Codec – AAC (Advanced Audio Codec)
    • Profile – Main@3.1
    • Bit Rate – ~55Kbps (VBR)
    • Channels – 1 (Mono)
    • Sampling rate – 44.1Khz

Enable Video Vs Enable Webcam for Participants

In Connect 9.2, there are two ways you can allow a ‘participant’ to turn their camera on – but each has slightly different results.

Enable Video – this option gives full rights to the video pod, same as a host. Or in other words, if you enable Video for any participant, that particular participant will get presenter rights over the Camera pod just like in case of Enhanced rights over any pod.

 

 

Untitled

Enable Webcam For Participants – this doesn’t give overall video pod rights (the participants can’t Force Presenter View, or choose who is in the main Filmstrip, etc.), but it allows them to turn their camera on. However you cannot be selective with this option – everyone will be able to turn on their camera.

Untitled1

FYI, If you “enable video” for a participant and then “enable webcam for participants” you lose the ability to “disable video” for any already enabled participant until you first “Disable Webcam for participants”

Be Aware of the Closed Captioning Pod Defaults

Last week we found out that Caption Colorado changed their IP address and port number for the Closed Captioning pod downloadable from the Connect Exchange Website. Here is the direct link to the Connect version 9 Closed Captioning Pod

The new Caption Colorado information includes:

If you are experiencing any trouble with the Closed Captioning pod while using it in a Connect Meeting with Caption Colorado, please set your host to “captionedtext.com” and to port 11100 in the adobe pod. Note that the new IP, 54.193.31.11, depending on your infrastructure’s network security settings, may need to be white-listed.

For an updated user’s guide referencing the Closed Caption Pod, see this PDF: http://platinum.adobeconnect.com/cc/

 

 

XML API Tips: Arkadin Profile Creation – Display Numbers

Previously I posted a blog entry on creating audio profiles via the XML API (http://blogs.adobe.com/connectsupport/xml-api-tips-creating-telephony-profiles-via-the-xml-api/).  This is an add-on article describing one additional step for finalizing an Arkadin profile.

Once you complete the Arkadin profile, you may notice the Conference Numbers aren’t showing up at the bottom of the profile (in the UI).  When you build a profile in the UI (without the API) and you save the profile, it will display the Conference Number and Conference Number Toll Free in a datagrid below the credentials.  When you create a profile with the API, it will not (unless you go into the UI and then click Edit and then Save again).

ark1

Notice no Conference Numbers listed in the grid below the Profile Status.

Also, if you create the profile with the API (and you don’t go into the UI and click Edit/Save), you may notice that the conference numbers are not displayed in the Dial In dialogue box in the meeting room for participants.  It will only list the Moderator and Participant codes.

ark3

Notice, no numbers appearing to call into…

To get the numbers to show up in the Profile and in the dial-in dialogue box inside of the meeting room itself, you need to add one additional web service call to your workflow as below:

Once you build your telephony profile from the previous article,  you need to take the profile-id value and make the ‘telephony-profile-dial-in-number-update‘ API call to input the new numbers as such:

/api/xml?action=telephony-profile-dial-in-number-update&profile-id=1379585623&location=Toll%20access%20number&conf-number=+1-8xx-xxx-xxxx&location=Toll%20free%20access%20number&conf-number=+1-xxx-xxx-xxxx

Where you add the profile-id
where you add all the ‘locations’ you want (the string for the conference number description)
where you add the applicable conference numbers for each (toll + toll free for instance)

After completing this step, if you view the profile in the UI again, you see the numbers appear:

ark2

Conference Numbers now listed in the grid below the Profile Status.

 

ark4

Notice now the number(s) appear.

Arkadin Audio Profile Conference Numbers

For Arkadin customers who integrate Arkadin audio profiles into Adobe Connect Meeting rooms, they need to be very careful in what numbers they are inputting into the profile fields when they are creating the telephony profiles.  Also, if these profiles are being provisioned automatically by an application utilizing the API, developers need to make sure the values they are passing in via the web services are also correct.

Arkadin has 3 phone numbers that are required when building an audio profile (UI pictured below).

arkprof

  1. Toll Access Number‘ (in the API this is: ‘x-tel-arkadin-conference-number‘)
  2. Toll Free Access Number‘ (in the API this is ‘x-tel-arkadin-conference-number-free‘)
  3. SIP Access Number‘ (in the API this is ‘x-tel-arkadin-conference-number-uvline‘)

It is crucial that you do NOT inadvertently put the Toll number (a non ‘1-8xx’ number) in for the Toll Free value and vice versa.  If you put a toll number in for the toll free number, the audio profile will save correctly, HOWEVER the UV line (Universal Voice) will not be able to connect to your meeting room when you try to start the audio (for Audio Broadcast and for Meeting Recording with Arkadin).  Universal Voice can only call out to a toll FREE number.  So if you are seeing your Arkadin audio conference not connecting correctly in the Adobe Connect Meeting room, please make sure to check your Arkadin profile that is assigned to the meeting, to make sure the toll free number is actually a toll free number, and the toll number is also correct.  The SIP access number should be set to the toll FREE number as well.

 

DB_PING_TIMEOUT Value Change

Recently we have discovered that a newer setting for on-premise (licensed) Adobe Connect servers may lead to a memory leak on the system in certain rare circumstances.  Here is some history and recommendations in case you believe you may be running into a memory leak problem in your Adobe Connect licensed environment and you are running a version newer than 9.0.3.

The DB_PING_TIMEOUT value was introduced back in Connect 7 (2008 timeframe).  It enables invalidated DB connections to be recognized quickly. In the absence of a reasonable value for this timeout, we have had instances in the past where critical CPS threads (e.g. the scheduler sweeper thread) have waited on a stale DB connection for too long, causing fastfails. This value had since always been set to ‘0’ which means there is no time out.  Since the default host health check time out value is 40 seconds, it is recommended that the DB_PING_TIMEOUT default value be set to 30 seconds, so that it is under the limit that causes potential server fast-fails. This was a fairly minor change in the config.ini, where the DB_PING_TIMEOUT value was changed from 0 to 30.  This was done at the Connect 9.0.3 version.  So every version above 9.0.3 will have the default set to 30.  [Important note – this value is in seconds, not milliseconds]

Recent longevity tests in version 9.2 suggested that this might be triggering a memory leak in the driver. The going theory for why that behavior wasn’t seen in previous longevity tests (between 9.0.3 and 9.2) is that we only upgraded to JRE 7 in 9.2. So the setting we were running with previously suddenly seemed to be a problem once we also upgraded to 1.7.

That value of 30 was introduced for a reason, so we don’t suggest turning it off without knowing that it causes a problem. On the Adobe hosted clusters, we have made the decision to do so since there were signs of memory issues even previously and we didn’t want to compound that.

That said, there are known issues with our driver and JRE 1.7, but only under some circumstances. In the case of Adobe Connect system administrators observing  (continuous) increases in heap memory usage, this parameter value should be set back to 0.

This can be done by changing this value either in the config.ini  from 30 to 0 (DB_PING_TIMEOUT=0)  or by adding this value in the custom.ini (it won’t be there by default, but if you add it, it will take precedence over what is in the config.ini)

 

XML API Tips: Creating Telephony Profiles Via the XML API

UPDATED – 4-11-2014

The workflow for creating telephony profiles for INTEGRATED telephony providers via the XML API has changed over the last year or so.  Here is an update on the supported method for creating telephony profiles for users using the Adobe Connect Web Services (XML API).

First, you need to find the telephony provider id number (provider-id) that you want to create a profile from.

To do this, you can make the ‘telephony-provider-list‘ API call as follows:

https://{connectURL}/api/xml?action=telephony-provider-list

The results will look like this (with obvious real values for the provider-id and acl-id parameters):

<results>
<status code=”ok”/>
<providers-account>
<provider provider-id=”xxxxxxxxx” acl-id=”xxxxxxxxx” provider-type=”integrated”>
<class-name>
com.macromedia.breeze_ext.premiere.gateway.PTekGateway
</class-name>
<adaptor-id>premiere-adaptor</adaptor-id>
<name>PGi NA</name>
<provider-status>enabled</provider-status>
</provider>
<provider provider-id=”xxxxxxxxx” acl-id=”xxxxxxxxx” provider-type=”integrated”>
<class-name>
com.macromedia.breeze_ext.telephony.Intercall.IntercallTelephonyAdaptor
</class-name>
<adaptor-id>intercall-adaptor</adaptor-id>
<name>InterCall</name>
<provider-status>enabled</provider-status>
</provider>
<provider provider-id=”xxxxxxxxx” acl-id=”xxxxxxxxx” provider-type=”integrated”>
<class-name>
com.meetingone.adobeconnect.MeetingOneAdobeConnectAdaptor
</class-name>
<adaptor-id>meetingone-adaptor</adaptor-id>
<name>MeetingOne</name>
<provider-status>enabled</provider-status>
</provider>
<provider provider-id=”xxxxxxxxx” acl-id=”xxxxxxxxx” provider-type=”integrated”>
<class-name>com.macromedia.breeze_ext.arkadin.ArkadinAdaptor</class-name>
<adaptor-id>arkadin-adaptor</adaptor-id>
<name>Arkadin</name>
<provider-status>enabled</provider-status>
</provider>
</providers-account>
</results>

Next, you take the provider-id from the call above, and make the ‘telephony-profile-update‘ call to create the initial telephony profile container.  The formatted call will look like this:

https://{connectURL}/api/xml?action=telephony-profile-update&principal-id=xxxxxxxxx&profile-status=enabled&provider-id=xxxxxxxxx&profile-name=xxxxxxxx

Where:
principal-id = the principal id of the user for whom you are creating the profile (obtained by other APIs).
profile-status=enabled (to enable the profile).
provider-id = the provider-id value from the first API call above, for which you are creating the profile.
 profile-name =the name of the profile you are creating for the user (it’s up to you for naming convention).

The results will look like this:

<results>
<status code=”ok”/>
<telephony-profile profile-status=”enabled” provider-id=”xxxxxxxxx” principal-id=”xxxxxxxxx” profile-id=”xxxxxxxxx”>
<profile-name>xxxxxxxxx</profile-name>
</telephony-profile>
</results>

Next, you take the applicable provider-id value from the result above, and run the telephony-provider-info API call to get the appropriate fields you would need to add to the profile (and hard code them into your app for future):

https://{connectURL}/api/xml?action=telephony-provider-info&provider-id=xxxxxxxxx

Where:
   provider-id = value from the first call.

The results will look like this (using Arkadin as an example).  What you really want to look for are the ‘telephony-provider-fields’ in the results.  The rest of the provider-dial-in-info can be ignored for the purpose of creating integrated profiles in this fashion.

The results in BLUE are the required ‘x-tel’ values for (in this example) Arkadin, to create a profile.

<results>
<status code=”ok”/>
<telephony-provider-fields>
<field provider-id=”xxxxxxxxx” field=”xxxxxxxxx” field-id=”x-tel-arkadin-conference-id” display-in-meeting=”none” required=”true” user-specified=”true” input-type=”text” is-hidden=”false”>
<name>Web login</name>
</field>
<field provider-id=”xxxxxxxxx” field=”xxxxxxxxx” field-id=”x-tel-arkadin-moderator-code” display-in-meeting=”hosts” required=”true” user-specified=”true” input-type=”text” is-hidden=”false”>
<name>Moderator pin code</name>
</field>
<field provider-id=”xxxxxxxxx” field=”xxxxxxxxx” field-id=”x-tel-arkadin-participant-code” display-in-meeting=”participants” required=”true” user-specified=”false” input-type=”text” is-hidden=”false”>
<name>Participant pin code</name>
</field>
<field provider-id=”xxxxxxxxx” field=”xxxxxxxxx” field-id=”x-tel-arkadin-conference-number” display-in-meeting=”none” required=”true” user-specified=”true” input-type=”text” is-hidden=”false”>
<name>Toll access number</name>
</field>
<field provider-id=”xxxxxxxxx” field=”xxxxxxxxx” field-id=”x-tel-arkadin-conference-number-free” display-in-meeting=”none” required=”true” user-specified=”true” input-type=”text” is-hidden=”false”>
<name>Toll free access number</name>
</field>
<field provider-id=”xxxxxxxxx” field=”xxxxxxxxx” field-id=”x-tel-arkadin-company-url” display-in-meeting=”participants” required=”true” user-specified=”false” input-type=”url” is-hidden=”false”>
<name>Other access numbers</name>
</field>
<field provider-id=”xxxxxxxxx” field=”xxxxxxxxx” field-id=”x-tel-arkadin-conference-number-uvline” display-in-meeting=”none” required=”true” user-specified=”true” input-type=”text” is-hidden=”false”>
<name>SIP access number</name>
</field>
</telephony-provider-fields>
</results>

Lastly, now you put everything together.  You take the profile-id that was from the newly created profile as part of the telephony-profile-update API above, and all the applicable ‘x-tel’ values from the telephony-provider-info call directly above, and piece them all together in an acl-field-update call as below (again, this example is for Arkadin):

https://{connectURL}/api/xml?action=acl-field-update&acl-id=xxxxxxxxx&field-id=x-tel-arkadin-conference-id&value=xxxxxxxxx&field-id=x-tel-arkadin-moderator-code&value=xxxxxxxxx&field-id=x-tel-arkadin-participant-code&value=xxxxxxxxx&field-id=x-tel-arkadin-conference-number&value=1-xxx-xxx-xxxx&field-id=x-tel-arkadin-conference-number-free&value=1-xxx-xxx-xxxx&field-id=x-tel-arkadin-conference-number-uvline&value=1-xxx-xxx-xxxx

Where:
 acl-id = the profile id value of the profile you created in the telephony-profile-update call above.
field-id = the x-tel params required by the provider.  As you can see above, for Arkadin, there are 6.
value = the value of the x-tel parameters.  You would get this from your conference provider.

The results will simply be:

<results>
<status code=”ok”/>
</results>

UPDATED – 4-11-2014
For completing an ARKADIN profile, please also add the following to the workflow:
http://blogs.adobe.com/connectsupport/xml-api-tips-arkadin-profile-creation-display-numbers/

For all the main providers on hosted, here are the required ‘x-tel’ values:

Arkadin:

x-tel-arkadin-conference-id
x-tel-arkadin-moderator-code
x-tel-arkadin-participant-code
x-tel-arkadin-conference-number
x-tel-arkadin-conference-number-free
x-tel-arkadin-conference-number-uvline

PGI:

x-tel-premiere-user-id
x-tel-premiere-password
x-tel-premiere-moderator-code

InterCall:

x-tel-intercall-participant-code
x-tel-intercall-leader-pin

MeetingOne:

x-tel-meetingone-conference-id
x-tel-meetingone-host-pin