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eZuce is pleased to announce the General Availability Release of uniteme and reachme 17.04.

As with the previous two releases, we’re continuing to focus more on fixes and minor improvements in uniteme as work continues on the next generation of code. Unite Web (the new User Portal) gains the ability to disable certain user functionality (this was also included in 16.12.1).

reachme gets a few new reports and improvements to situations where users get their web browser disconnected from the reachme server. In previous versions, if an agent had their browser disconnect from the server the call would be requeued. Now on browser disconnect, the system admin has the option to let a phone call continue instead of the only option being to requeue the call.

Also as always, thanks to the Dev & QA team at eZuce for their excellent work on this release.

In all 55 issues (enhancements / fixes) are addressed for uniteme and reachme in this release.

The next uniteme and reachme release will be 17.08.

Highlights

uniteme New Features:

  • Unite Web Admin Control over User Features

  • Unite Web user control over Conference Bridge Entry / Exit tones

  • Unite Web user control over Conference Bridge Voice Announce of Entry / Exit

  • Statistics collection for sipXproxy service (phase 1 of a multi-part project)

uniteme Improvements:

  • Improvements to Yealink phone configurations

  • REST API to create/modify a user/user group and set properties

  • Improved backup speed by breaking apart backup and compression

  • Improved CDR display in Unite Web for users. Added duration and ability to select Time Zones.

  • Make Unite Web tab blink when there’s a new message.

reachme New Features:

  • Improve reachme agent reconnect in case of network loss - leave call up

  • Added ‘agent’ specific skill to remove skill in a recipe action.

  • New Reports:

    • Voicemail Detail report

    • Voicemail Overview report

    • Agent States report

reachme Improvements:

  • Ensure that reachme standard and custom reports are Backed up with system backup process.

  • Report Improvements:

    • Added magic skills to the Agent Availability report

    • Added ‘Sent’ to the Agent Group Activity report (total CDRs with direction outbound and disposition of agent initiated)

Notes

  1. Full Release Notes with installation information are located here: 16.12 Full Beta Release Notes

Who Should Install?

This release is recommended for all 4.6 and later installations. If you have a patch installed to your system a new patch may be required. Please contact sa@ezuce.com if think you may have a patch applied as that may be replaced during the update.

eZuce's software products continuously progress through an Agile based development methodology that keeps feature functionality comprehensive and up-to-date in response to evolving market and customer requirements.

New software releases are made at a rate of four to six releases a year. Releases are numbered in the "<yy>.<mm>.<uu>" format where <yy> and <mm> designate the year and the month, respectively, in which a release is made generally available. Where applicable, <uu> corresponds to an update release relative to a general release on which fixes are made available.

In order to ensure service continuity and stability, customers may keep their production environments unchanged for up to a 6-month period during which release updates or patches are made available. After a release is more than 6-months old, eZuce customers would have to upgrade to the latest generally available release - inclusive of all fixes to date and any new patches.

Questions

If you have questions about updating you can email sa@ezuce.com or if you need assistance with the update please contact your account manager or email sales@ezuce.com.

Software Release History

 

We're currently running on a 4-month release cycle.

  • April release for 2016 is 16.04
  • August release for 2016 is 16.08
  • December release for 2016 is 16.12
  • April release for 2017 is 17.04
 

Release Level History

  • 14.04   - April 30, 2014
  • 14.04.1 - June 01, 2014
  • 14.04.2 - July 11, 2014
  • 14.04.3 - October 24, 2014
  • 14.10 - February 5, 2015
  • 15.04 - April 29, 2015
  • 15.05 - May 27, 2015
  • 15.06 - June 30, 2015
  • 15.08 - August 31, 2015
  • 15.10.1 - December 9, 2015
  • 15.12 - January 6, 2015
  • 16.02 - March 14, 2016
  • 16.04 - May 31, 2016
  • 16.08 - October 6, 2016
  • 16.12 - January 17, 2016
  • 17.04 - April 18, 2017

System Requirements

For a reasonably performing system, we recommend the following configuration.

Minimum hardware requirements

  • Pentium 4 or Xeon processor @ 2.0 GHz Core 64bit or higher
  • Minimum 4 GB of RAM with sufficient swap space
  • 80 GB disk (75 users depending on usage patterns)

Notes:

  • uniteme supports an unlimited number of voicemail boxes, the total number of hours of recorded messages is determined by the size of the hard-disk. As a rule, for every minute of recorded messages, you will need 1 MB of disk space (About 3 hours per 10 GB of disk space).
  • reachme requires more memory, processor and disk space. Please consult with eZuce SA team for your specific installation.

Operating System

CentOS/RHEL 6 x86_64 with latest updates is required.

Devices

Phones

  • Polycom VVX Devices with firmware 5.2.5 (split) are recommended for new installations
  • Polycom SoundPoint IP Devices should run firmware 4.0.7 (split)

Gateways

  • AudioCodes Gateways are recommended for PSTN connectivity

SBCs

  • Frafos, Sangoma, AudioCodes, Acme Packet and Ingate SBC's are recommended for SIP Trunking and Remote Worker connectivity (commonly referred to as sipXbridge and MediaRelay services respectively).
  • NOTE: The eZuce uniteme - "Use built-in SIP Trunk SBC" found in Gateway Details for use with Trunking or Remote Worker solutions should be used only for lab purposes. The openUC "Built-In SIP Trunk SBC" (sipXbridge) will not be supported in any production or live environment. Additionally, sipXbridge does not work in an HA environment.

Documentation

Technical Reference Manuals, User Guides, Reach Reference Manuals, and other technical and user information can be found under the following link: Documentation Page

Installation and Upgrade Notes

Installation note

After uniteme 17.04 is downloaded and installed, the clusterId read tag is unique (same as locationId). Follow these steps to propagate the new read tags to the MongoDB replica set:

  1. In the uniteme menu, click System>Database.
  2. Click the Add query metadata button.
  3. To verify that the MongoDB replica contains the unique read tags, run from the command line:
//mongo
rs.config();//

Special MongoDB note

Please be aware of these Mongodbrequirements http://docs.mongodb.org/manual/reference/ulimit/ Note: Both the “hard” and the “soft” ulimit affect MongoDB’s performance. The “hard” ulimit refers to the maximum number of processes that a user can have active at any time. This is the ceiling: no non-root process can increase the “hard” ulimit. In contrast, the “soft” ulimit is the limit that is actually enforced for a session or process, but any process can increase it up to “hard” ulimit maximum.Every deployment may have unique requirements and settings; however, the following thresholds and settings are particularly important for mongod and mongos deployments:

ulimit –a
-f (file size): unlimited
-t (cpu time): unlimited
-v (virtual memory): unlimited
-n (open files): 64000
-m (memory size): unlimited
-u (processes/threads): 32000
 

Always remember to restart your mongod and mongos instances after changing the ulimit settings to make sure that the settings change takes effect.If you limit virtual or resident memory size on a system running MongoDB the operating system will refuse to honor additional allocation requests. After every install/upgrade please check that "cat /proc/$pid_of_mongo/limits" have the recommended value of 655350. To make this value permanent you need to create this file /etc/security/limits.d/99-mongodb-nproc.conf and add the following lines:

mongodb soft nproc 64000
mongodb hard nproc 64000
mongodb soft nofile 64000
mongodb hard nofile 64000

Special Patch Note

If you have a patch installed to your system a new patch may be required. Please contact sa@ezuce.com if think you may have a patch applied as that may be replaced during the update.

Installing from ISO image

Download uniteme ISO

Download the ISO image corresponding to your hardware and write the image to a DVD.

Install uniteme

  • Boot from the DVD created with the uniteme ISO image.
  • Press Enter at the boot screen below to begin the uniteme installation.
  • Select Manual Configuration under Enable IPv4 support and select OK.
  • Set a static IPv4 address with the corresponding networking information and click OK.
  • In certain situations, a warning of the use of indicated storage devices will be displayed.
  • Select the language to be used during the installation.
  • Select the keyboard layout to be used.
  • Select the timezone to be used.
  • Set a root password.
  • Login to the system as root with the password you provided earlier and continue on to the Configure of uniteme.

Installing from Repository

uniteme can be installed using the following procedure

  • Download and install CentOS 6.x minimal ISO
  • Run the following command:
yum update && reboot
  • Run the following commands to retrieve and run the eZuce uniteme installer:
curl https://download.ezuce.com/openuc-setup > /usr/bin/openuc-setup
chmod +x /usr/bin/openuc-setup
openuc-setup

This utility will guide you through the process of installing uniteme from the eZuce software repository.

Upgrade from previous versions

Modify the repo file in /etc/yum.repos.d and replace the baseurl= with the location of the repository you'd like to upgrade to.

Identify any existing 'rpmnew' or 'rpmsave' files on the system with:

find / -print | egrep "rpmnew$|rpmsave$"

 

As root, execute the following commands:

yum clean all
yum update

Note any additional 'rpmnew' or 'rpmsave' files that may have been created by running find command again 

find / -print | egrep "rpmnew$|rpmsave$"
 

If there are any files that didn't get overwritten by yum, please see 'Modified Files Upgrade Note' information below.

A system reboot after the update has completed is recommended.

SEC Service Upgrade Note

When upgrading uniteme from openUC 4.6 Update 11 or 14.4.3 to 15.06 follow these steps to ensure the SEC service is correctly running:

  • 1. Upgrade from 4.6 Update 11 or 14.4.3 to 15.06.
  • 2. After the upgrade is complete, perform the usual restart.
  • 3. Once possible, connect via CLI and monitor processes using top. Notice that the SEC process is using a lot of CPU memory.
  • 4. Perform another restart OR restart only the Sipxlogwatcher service.

Modified Files Upgrade Note

If you have manually modified any system related files or some files are not as yum would expect them to be, the yum update process may not overwrite them. It will instead create 'rpmnew' or 'rpmsave' files and not overwrite the files. The administrator may have previously modified the files knowingly or as part of a patch supplied by TAC.

To check your upgrade.log and search for *.rpmnew *.rpmsave on your system check the upgrade log:

You will be responsible for merging any changes from the old file to the new or contacting Technical Support if you require assistance.

Support Tips and Contact Information 

Please see the Getting Support section for support tips and support contact information

Issues Sorted by Issue Number

 JIRA nameRN ContentEnhancement/Fix/Known IssueKey words
SIPX-526Missing DNS record for Proxy-Forwarding to RegistrarFix for DNS records when a server did not have Registrar enabled.

To reproduce issue:
Create a cluster with 3 Servers in the following configuration:
PBX01 with Proxy and Registrar
PBX02 with Proxy and Registrar
PBX03 with Proxy

With this setup every message, sent to the PBX03-Proxy won't get to one of the existing Registrars. Results in lost calls

Config on PBX03-Proxy routes to rr.pbx03.voip.domain.de
This name is not generated in DNS

Current DNS config is:

_sip._tcp.rr IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx01 IN SRV 10 10 5070 pbx01
_sip._tcp.rr.pbx01 IN SRV 30 10 5070 pbx02
_sip._tcp.rr IN SRV 30 10 5070 pbx02
_sip._tcp.rr.pbx02 IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx02 IN SRV 10 10 5070 pbx02

DNS should be configured like this to make it work:

_sip._tcp.rr IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx01 IN SRV 10 10 5070 pbx01
_sip._tcp.rr.pbx01 IN SRV 30 10 5070 pbx02
_sip._tcp.rr IN SRV 30 10 5070 pbx02
_sip._tcp.rr.pbx02 IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx02 IN SRV 10 10 5070 pbx02

_sip._tcp.rr.pbx03 IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx03 IN SRV 30 10 5070 pbx02
FixDNS
SIPX-539Yealink Emergency DND FeatureEnhancement to allow provisioning support for Yealink's Emergency DND Feature.

From Yealink Provisioning Guide:

Specify the authorized numbers when DND is enabled.
Parameters:
features.dnd.emergency_enable
features.dnd.emergency_authorized_number
EnhancementYealink
SIPX-540Yealink Call Number FilterEnhancement to allow provisioning support for Yealink's Call Number Filter.

From Yealink Provisioning Guide:

Configure the characters the IP phone filters when dialing.
Parameters:
features.call_num_filter
EnhancementYealink
SIPX-560Alert InfoExternalThis is an improvement of Proxy Plugin to set Alert-Info-Header.

Some phones (e.g. Yealink) bypass the Proxy if the From-Header do not end with @<sipdomain>.

For SIP-Devices that have no ability to add custom headers to a SIP Message (e.g. Patton), it is necessary to scan the from the header for a tag (x-sipx-alert-info=external) to set the Alert-Info Header for From-Uris with the SIP Domain inside.
EnhancementYealink
SIPX-565Yealink Provisioning of Voice Quality MonitoringEnhancement to the Yealink provisioning to enable Yealink configuration of RTCP-XR parameters.

Yealink can send a Report to a data collector to get information about the quality of the last call.

Event is: vq-rtcpxr

On activation, the Phone sends a Publish Message with the Report to a service you can configure
EnhancementYealink
SIPX-569Syslog only receiving on UDP 514In older sipXcom releases, the default Syslog transport setting for Polycom phone groups was UDP. In release 16.02, the default setting syslogtransport setting is TCP, which triggers the phones to send log file information to Sipxcom using TCP transport port 1468.

This is a fix to setsyslog transport back to UDP which is preferred over TCP to preserve TCP sockets.
FixsipXconfig
SIPX-572Jitsi Provisioning DNDAn enhancement to Jitsi provisioning with the following parameter:

net.java.sip.communicator.impl.protocol.RejectIncomingCallsWhenDnD

For sipXcom/Uniteme it is necessary to set this parameter has to be set to "true", because the Proxy/Registrar could not handle the DND itself.
EnhancementJitsi
SIPX-577DNS NAPTR prefer TCPEnhancement to adjust the DNS NAPTR records to have client prefer TCP vs. UDP since TCP is the preferred protocol for SipXcom/UniteMe

Current DNS NAPTR configures them to be equal

voip.domain.de. IN NAPTR 2 0 "s" "SIP+D2U" "" _sip._udp
voip.domain.de. IN NAPTR 2 0 "s" "SIP+D2T" "" _sip._tcp

This should be changed to the following so if some devices use auto configuration, TCP will be chosen.

voip.domain.de. IN NAPTR 2 0 "s" "SIP+D2U" "" _sip._udp
voip.domain.de. IN NAPTR 1 0 "s" "SIP+D2T" "" _sip._tcp
FixDNS
SIPX-578Stats collecting submodule for ProxyEnhancement to sipXproxy that is required to implement internal metrics collecting submodule for proxy and output collected metrics into file with pre-configured interval.

In the first approach, Proxy shall have statistics file with name=value content (like "ProxyQueueSize=123"). Single line for a single metric.
The file should be updated every StatsUpdateInterval seconds configuration parameter. The default value will be 15 seconds. In the future, if we will need we can add this to WebUI configuration.
EnhancementsipXproxy
SIPX-581Update Admin GUI w/Blog Posts and LinksThis is an enhancement to the Admin GUI to receive blog post updates from sipxcom.org website in the Admin GUI as well as links to important sipxcom.org URLs.

Add RSS feed capabilities to Admin GUI. The feed should come from http://sipxcom.org/feed/

Add the important links on the right as shown in Mockup:

Downloads (ISO's & RPM's): sipXcom ISO Images and RPM Repositories
(link the text after the ':' to http://wiki.sipxcom.org/display/sipXcom/sipXcom+ISO+Images+and+RPM+Repositories)
JIRA Issue Tracker: http://jira.sipxcom.org
sipXcom on Github: https://github.com/sipXcom
sipXcom User's Mailing List: https://groups.google.com/d/forum/sipxcom-users
sipXcom Developers Mailing List: https://groups.google.com/d/forum/sipxcom-dev
Paid Version & Optional Features: https://www.ezuce.com

At top of the page darken icons to 50% gray. Add Paid Support, change 'Help' to 'Docs'.

Paid Support should link to: http://ezuce.com/products-solutions/procare-sipxcom-support
EnhancementsipXconfig
SIPX-585Yealink Resource List SubscriptionFix for Yealink Provisioning to check if lines on the phone have BLFs.

Current Issue:
If a user has no BLFs, Yealink start to spam to proxy with resource list subscriptions and if you have enough Yealink to proxy stops working.
FixYealink
SIPX-587Rotate proxy_stats.json file - proxy statsThis is an enhancement in support of the proxy statistics enhancement. The file where proxy stats are collected should be included in the Logrotate mechanism.EnhancementsipXconfig sipxproxy
SIPX-600Custom settings enhancement for Yealink provisioningEnhancement to add the ability to add custom settings to Yealink configuration.

This is a good feature with Polycom provisioning which is now available for Yealink provisioning. Settings which are currently missing in GUI could be set via this custom config.
EnhancementYealink
UC-2879Allow IVR to set a Reach Agent Specific Skill

Enhancement to allow the IVR to set an agent specific skill for a call.

Currently, an IVR script can set skills required for a Reach call prior to sending the call along to Reach. This is done by setting the "skills" variable via a command such as:
skills = "skill1,skill2,skill3";
session:setVariable("skills", skills);

Each of the skills "skill1", "skill2" and "skill3" would then be built in the Reach configuration and those calls coming from the IVR would require those skills.

Note that the default weight of this call should be higher than other calls by default such that the agent isn't offered a regular queue call prior to being offered this agent specific call. This would be similar to how a transfer to agent works (internally treated with higher priority).

It is desirable to allow the IVR to set an agent specific skill for a call as well. This should be done in the same way as setting other skills.

Use Case:
1. caller calls into IVR
2. IVR collects a ticket number
3. IVR dips a back end system to see what agent is assigned to this ticket
4. IVR builds skill list with base skills such as "ServiceDesk", "Tech", etc AND the agent specific skill that is assigned to the ticket.

EnhancementReachme
UC-3448SAA does not apply some configuration file parametersFix for 2 configuration file parameters which were not applied to SAA service, so SAA would always work with its defaults for this params:
server-min-expires (300 secs)
server-default-expires (3600 secs)

Steps to reproduce:
1. Change the values of the 2 params in the openuc-saa.ini file (i.e. change them to 60)
2. Start SAA with INFO log level
3. Grep after the "expire" message in openuc-saa's log file:
qq
"2014-12-12T15:53:48.977944Z":50:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSClientSubscriptionHandler::SSSClientSubscriptionHandler::0x7fffc3bbfe10 Client subscription default expire seconds = 3600"
"2014-12-12T15:53:48.978167Z":56:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerSubscriptionHandler::SSSServerSubscriptionHandler::0x7fffc3bbfee0 Server subscription min expires = 60"
"2014-12-12T15:53:48.978197Z":57:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerSubscriptionHandler::SSSServerSubscriptionHandler::0x7fffc3bbfee0 Server subscription default expires = 60"
"2014-12-12T15:53:48.978219Z":58:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerSubscriptionHandler::SSSServerSubscriptionHandler::0x7fffc3bbfee0 Server subscription max expires = 86400"
"2014-12-12T15:53:48.978353Z":64:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerBase::readConfigOptions::0x7fffc3bbf980 min expires = 300"
"2014-12-12T15:53:48.978383Z":65:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerBase::readConfigOptions::0x7fffc3bbf980 default expires = 3600"
qq
As can be seen, the SSSClientSubscriptionHandler::SSSClientSubscriptionHandler() and SSSServerBase::readConfigOptions() functions are using the default values: 300 and 3600, respectively.

Also, the values can also be checked at runtime: client subscriptions are always created using the default Expire (i.e. 3600) and the minimum expiration (which can be seen when SAA refreshes the SUBSCRIPTION during a call) is always the default one (i.e. 300)
FixSAA
UC-3573"This week" queue statistics seen by Supervisor are cleared on TuesdayFix for "This week" queue statistics which were being cleared on Tuesdays.

To reproduce:
1. On Monday - Place a few calls to make sure the Queue statistics get populated.
The Queue statistics have been updated ok for all of the options (Last 15 Minutes, Issue :Today .. This Month etc).
2.Next day, Tuesday, check the statisIssue:

Statistics showed that "This Week" statistics was 0. (as if yesterday was Sunday... and the week ended)

Expected: Statistics to show the number of calls placed Monday.

Place a few more calls and "This week" will get populated with statistics based on calls that were just placed.
FixReachme
UC-3683REST API to create/modify a user/user group and set propertiesEnhancement to allow for user management through a rest API.

Currently, we have SOAP support for user creation, but there is no support on user settings (like user called id for example)

Use the new REST Api engine based on Apache CXF to create such api. The settings management is generally handled by current existing REST support
EnhancementsipXconfig
UC-4061Add Agent States Reach ReportEnhancement to add a Reachme report with the following details.

Inputs:
Date/Time, skills list (multi-select; default that to all skills)

Outputs:
Agent name, Login, Current state during the input date/time

Additional Output
A separate table formatted output with a summary of the data shown in the details section defined above. This summary section will have columns headers of ...
Available, Ringing, On Cal, Wrap Up, Outbound, Other, Release1, Release 2 ... Release "n" (an entry for each release reason built in the system)
Data under these headers will be a total count of agents that were in that state
EnhancementReachme Reports
UC-4141Ensure Reach Reports are backed up with the backup process - standard and customEnhancement to ensure that all standard deployed Reports, as well as any custom Reports that may have been deployed, are backed up as part of the system backup.

Obviously, underlying data used by the Reports should also be backed up.

Restoring this backup should result in getting all of the standard Reports, still having any custom Reports that were previously created and data required to run the Reports.
EnhancementReachme Backup
UC-4146Update Agent Group Activity Reach Report to have Outbound Sent ColumnEnhancement to the Agent Group Activity report to add Outbound Sent column.

Update this report as follows:

Under the Outbound subsection;
Change total outbound column name to Started (total CDRs with direction outbound)
Add column called Sent under the Outbound activity area (total CDRs with direction outbound and disposition agent initiated)
EnhancementReachme Reports
UC-4201Add a Voicemail Overview ReportFixed an issue with the Reachme portal reconnection mechanism where it would appear to work if you used an IP instead of the server name, but the server connection wou, in fact, t be disconnected.

To reproduce the issue:
If you insert the server's ip address e.g. 10.5.0.210/reach to get to the dashboard, then you would normally get automatically redirected to the "server_fqdn"/reach.
But then if you insist and edit that address, and replace the fqdn with the ip address, then, in this case, it allows you to use it and it does not redirect you any more to server's FQDN.

In this case, after you login and a network disconnection occurs (larger then 15s), the dashboard will "appear" to work but in fact it won't.
If you click the Release button, it gets grayed out.
If you place calls in the queue, the agent is not alerted.

Agent must refresh the page to make it work again.
EnhancementReachme Reports
UC-4202Add a Voicemail Detail ReportEnhancement to add a new report that offers detailed information on Voicemail activity. This is the same as UC-4201 except with an additional input parameter of interval and the output is grouped by the interval input. Interval should default to 60 minutes.EnhancementReachme Reports
UC-4207Agent State History report shows "null" when selected agent has no activityFix to solve the case where the report shows "null" when a selected agent has no activity.

To reproduce
1. Create a new agent under some agent group
Don't do anything with this agent
2. Login with a supervisor, go to Reports, select Agent State History report and the correct timeframe.
Select the agent group , and then choose the newly added agent
3. Generate report
Issue: report shows "null"
FixReachme Reports
UC-4228Add agent specific skill to remove skill recipe actionEnhancement to add a "special entry" in the list of skills to possibly remove in the remove skills recipe action.

Currently, when building a remove skills recipe action in the Reach admin, the skills listed in the action box that you can remove include the following:
All skills that the admin has added to the system
Magic skills of:
Node
Queue
All
Client

This proposed feature is to allow the admin to remove any agent specific skill.

Admin UI should display a new value under the Magic subsection. This new value will be "Agent".

If selected, when the recipe step is executed based on the criteria of the recipe step being matched, Reach should remove any/all agent specific skills from the call.

This relates to calls that have been transferred to a specific user. In such a case, there is an agent specific skill added to the call when it is transferred.

More importantly, it relates to UC-2879 which adds the ability for an ivr script to add an agent specific skill. Once that is added, then we need to have the ability to remove this agent specific skill from the call based on a recipe step.

The use case is as follows:
Call comes into IVR
IVR interacts with the caller and with a back end ticket system
IVR finds that there is an open ticket in the ticketing system
IVR finds the "ticket owner" from the back office system (e.g agent 2222 is the owner of the ticket)
IVR sets brand, queue and skills on call; skills include agent specific skill of 2222
IVR sends call to reach queue

Now, reach will only deliver the call to the specific agent 2222 since that agent specific skill 2222 is only assigned to agent 2222
At this point, you would then want to be able to remove this skill so that the call could go to other agents (e.g. agent 2222 is not logged in).
EnhancementReachme
UC-4284Add magic skills to Agent Availability reportEnhancement to the Agent Availability report to also show the Magic Skills.EnhancementReachme Reports
UC-4290Enhancement request for Unite Web CDR durationAn end user would like to see a call Duration column (or, alternatively, the End Date/Time) in the end user Call Detail History in Unite Web.EnhancementUnite Web
UC-4296Web UI Search searches Voicemail PIN TokenFixed an issue where Voicemail PIN was included as a searchable field when searching for an extension in the system.

To reproduce the issue:
Example: You search for "96"

The search will show you every user starting with 96 and user with a voicemail pin token that starts with 96
FixsipXconfig
UC-4297Increase elasticsearch user limitsFix for the Elasticsearch user file limits which are set too low for larger systems.

We should have an entry in /etc/security/limits.d/ for user elasticsearch to increase the current number of open files and processes that user can spawn per this description: https://www.elastic.co/guide/en/elasticsearch/guide/current/_file_descriptors_and_mmap.html
FixsipXconfig
UC-4307Add new settings for User Portal configurationThis is an enhancement in support of Unite Web work to allow an Administrator to be able to control the enabling and disabling of features of the User Portal by User and by User Group.

This work is in support of the 4 new features.

Feature 1 - Disable Dial Pad and Search icons
These options would be configurable in the Uniteme Administration GUI in Users -> Users -> “username” or Users -> User Groups -> “user group name”. In the left side menu there would be a new menu item called User Portal.

In the User Portal configuration page there would be the following configuration options for this feature:
Enable Dial Pad Icon Type: Checkbox Default: Enabled
Enable Search Icon Type: Checkbox Default: Enabled

Feature 2 - Disable Contact Click to Call and Chat
In the User Portal configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Contact Click to Call Type: Checkbox Default: Enabled
Enable Contact Click to Chat Type: Checkbox Default: Enabled

Feature 3 - Disable Click to Call from Conf Bridge
In the User Portal configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Conference Bridge Click to Call Type: Checkbox Default: Enabled

Feature 4 - Disable Unite Web menu items
In the Unite Web configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Activity List Type: Checkbox Default: Enabled
Enable Contacts Type: Checkbox Default: Enabled
Enable Group Chats Type: Checkbox Default: Enabled
Enable Conference Bridge Type: Checkbox Default: Enabled
Enable Voicemails Type: Checkbox Default: Enabled
Enable My Profile Type: Checkbox Default: Enabled
Enable Call History Type: Checkbox Default: Enabled
Enable Settings Type: Checkbox Default: Enabled
Enable Settings Personal Attendant Type: Checkbox Default: Enabled
Enable Settings Call Forwarding Type: Checkbox Default: Enabled
Enable Settings Speed Dials Type: Checkbox Default: Enabled
Enable Settings User Settings Type: Checkbox Default: Enabled
Enable Settings User Settings Change Password Type: Checkbox Default: Enabled
Enable Settings User Settings Voicemail PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Announcement Type: Checkbox Default: Enabled
Enable Settings User Settings eMail Type: Checkbox Default: Enabled
Enable Settings User Settings Attach audio Type: Checkbox Default: Enabled
Enable Settings User Settings Alternate eMail Type: Checkbox Default: Enabled
Enable Settings User Settings Alternate Attach audio Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Room Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Enabled Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Name Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Moderator PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Participant PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Max. members Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Quickstart Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Auto-record Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Moderated Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Public Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Entry Tone Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Exit Tone Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Voice Announce Entry Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Voice Announce Exit Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Audio source Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Personal MoH Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Files Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Audio file Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Conference enter Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Conference exit Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Voicemail begin Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Voicemail end Type: Checkbox Default: Enabled
Enable Settings Sound Notifications Type: Checkbox Default: Enabled
EnhancementUnite Web sipXconfig
UC-4311Polycom SoundPoint IP 650 and 560 background images are not workingFixed an issue with background images on Polycom SPIP 650 and 560 phones.FixPolycom sipXconfig
UC-4313Conference settings REST API to include new parametersThis is a Rest API enhancement to support Unite Web and Unite Lite enhancements.

We need to add new parameters:
Play Entry Tone Type: Checkbox
Play Exit Tone Type: Checkbox
Play Voice Announce Entry Type: Checkbox Default: Enabled
Play Voice Announce Exit Type: Checkbox Default: Enabled

to existing REST API
method: 'GET' and 'PUT'
/my/conferences/"conference name"
EnhancementUnite Web Conferencing sipXconfig
UC-4315SSS crash on empty "Refer-To" fieldFixed an issue with SSS where it would crash if it got messages with empty Refer-To field.FixSSS
UC-4316Improve Reachme agent reconnect in case of network loss - leave call upEnhancement to the Reachme agent/supervisor portal for reconnecting in the event of network connectivity loss.

In the current situation if the agent loses network connection to the Reach server, the reconnection mechanism will detect this connection loss and try to recover from it in two ways:
1. If the drop is less than 17 seconds in length the agent does not get released or logged out and his call only loses voice path for these few seconds, call not getting dropped.
2. If the drop is bigger, then the agent is prompted that connection to the server is lost, the agent is released, and the call is dropped in approx 40 seconds.

However it seems our customers would like to have this feature somehow improved.
Customers are saying: "The Reach web client has a very tight connection with a very low tolerance for connections blips, making it uniquely fragile in situations like this, causing not only the calls to drop or re-queue, but all the agents to release."

Some tests we've executed :
https://docs.google.com/spreadsheets/d/1HHdFH6FD6e86qcErwG_mOnf09XHGn2020KjTdef0Cwg/edit#gid=0

CURRENT PLAN FOR FIX IS AS FOLLOWS:
1. If the websocket connection is detected from UI side, attempt to reconnect as usual. This will include the new fix for the issue where a reconnect would not be successful unless the URL was an exact match to the standard reach URL.
2. If the ReachMe server side recognizes a WebSocket disconnect, it will
- leave the call up
- not attempt to use the socket connection any further
- show the agent in the agent manager as still being on a call
- record complete CDR for the call as it does for any other normal call
- Then, at the end of the call, kill off the agent connection.
3. If the UI socket connection is lost and the agent still has an active call, the agent or the system may reconnect. If they do while the call is still active, they will receive a warning that reconnecting the UI session will end the active call. If they proceed, the call will be hung up (just like any other agent initiated hang up) and the UI session will be started new.
EnhancementReachme
UC-4324Unite Web call history improvementsEnhanced the Unite Web Call History.

These are some of the changes that need to be implemented:
1.show timezone drop down as in old style portal, but defaulting to show the timezone of the user's PC (old style portal relies on what is set under User->Time Zone, but we don't need that)
2.Show call history entries by default, based on the default selection.
As soon as you go on the call history page, without hitting Apply.
3.Reverse Apply button location with To/from box
4.When using To/From box, results should show up if you both hit Enter or click Apply
5.Fix time format to 00:00:00 in Start/Stop columns instead of 00:0:00
6.Sorting of columns in results, if it's easy
EnhancementUnite Web
UC-4328REST API to manage user properties for user portalAn Enhancement to support the work for Unite Web work. Provide set of rest APIs that a regular user (USER_ROLE) can access, in order to update logged in user properties/settingsEnhancementUnite Web sipXconfig
UC-4330When agent connects via IP address instead of FQDN, reconnection mechanism does not workFixed an issue with the Reachme portal reconnection mechanism where it would appear to work if you used an IP instead of the server name, but the server connection would, in fact, be disconnected.

To reproduce the issue:
If you insert the server's ip address e.g. 10.5.0.210/reach to get to the dashboard, then you would normally get automatically redirected to the "server_fqdn"/reach.
But then if you insist, and edit that address, and replace the fqdn with the ip address , then in this case it allows you to use it, and it does not redirect you anymore to server's fqdn.

In this case, after you login and a network disconnection occurs(larger than 15s), the dashboard will "appear" to work but in fact it won't.
If you click the Release button, it gets grayed out.
If you place calls in the queue, the agent is not alerted.

The agent must refresh the page to make it work again.
FixReachme
UC-4343Reachme Reports section does not display the full set of input controls - missing scrollbarFixed an issue that was presented when a report had multiple input controls. When this was the case, the display window was not large enough. The solution is to implement a scrollbar for the window.FixReachme Reports
UC-4349Separate backup tar & compressionAn administrator would like to speed up the backup process.

For backup scripts, separately specify the tar and the gzip and set the compression level to 1.

first: tar -cvf
then: gzip -1

This will increase speed but also increase the backup size (but probably not significantly. More concerned with the backup window on larger systems than backup size on smaller systems.
EnhancementBackup
UC-4354Allow Reach Admin to Define Call Recovery MechanismFixed an issue where users of Unite Web were not getting unread messages they may have been sent while they were offline.

Reproduce Issue:
Two users must exist on system.
1. Login with user 1 on Firefox/Chrome Windows in Unite web
2. Do not login with user 2 (yet)
3. Leave some offline messages to user 2 from user 1.
4. Login with user 2 and make sure you are under Activity List

Issue: After the second user logs in, there is no notification message that informs the user that are some unread messages.
This should show up on the chat entry in Activity List.
Currently, you see the message user 1 sent (the order in which they show up is wrong UW-363), but you will not know they are new because of the missing notification.

Reproduces on all platforms
EnhancementReachme
UC-4355Change Reach call recording archival to be more intuitiveThere is a current setting in the Client record in the reach configuration called 'Days to Retain Recordings in Archive'. This value CURRENTLY represents the total number of days that a recording has been in the system before it should be removed from the archival location.

An example of how it was working:
A call comes into the system on 2/1 and is set to be recorded and kept in the system locally for 5 days and then be archived. The 'Days to Retain Recordings in Archive' is set to 10. The recording will be made and placed into the local system for 5 days. It will then me moved to the archive and reside there for an additional 5 days (total of 10).

This Jira is a request to change that the archiving back end operates in the same manner as the GUI says it will.

An example of how it should work after this fix:
A call comes into the system on 2/1 and is set to be recorded and kept in the system locally for 5 days and then be archived. The 'Days to Retain Recordings in Archive' is set to 10. The recording will be made and placed into the local system for 5 days. It will then me moved to the archive and reside there for an additional 10 days (total of 15).
FixReachme
UC-4363Jitsi provisioning automatic display nameAn administrator would like such that if nothing is entered for the Display name, that the provisioning fills this with name and surname stored in the user profile by itself (same behavior as Polycom Plugin).

The current Version of Jitsi Provisioning can configure the line display name (Lines > SIP > Displayname), it's just not utilizing the name and surname from the user profile.
EnhancementJitsi
UC-4376Disable MWI subscription if Voicemail permissions are disabledFixed an issue where Polycom phones are trying aggressively to subscribe for Event: message-summary.
To those Subscribes proxy returns 403 but the server is flooded by SUBSCRIBES by the users that have Voicemail permissions are disabled.
FixPolycom
UC-4389Agent State History/Agent Unanswered Call Details missing when upgrading from 16.04.stage/16.08.stage to 16.12.stageFixed an issue with Agent State History and Agent Unanswered Call Details missing when upgrading from 16.04.stage/16.08.stage to 16.12.stage.
Also the Agent Group prod & Agent Producitivity by Group fail to open.
FixReachme
UW-355On Safari and IOS, the scrolling is not working properlyFixed an issue under the Profile tab that was reproduced after the user tried to edit some fields and then scroll down and save it. After saving, when the user would scroll up to the beginning the user would notice that the scroll is working very hard and sometimes is not working at all (because the user can accidentally scroll from the margin and as result entire web page will be scrolled).

If the user scrolls up or down from the middle of the screen then it will work ok without problems.
FixUnite Web
UW-362No new message notification in Activity List for unread messages received while offlineFixed an issue where users of Unite Web were not getting unread messages they may have been sent while they were offline.

Reproduce Issue:
Two users must exist on system.
1.Login with user 1 on Firefox/Chrome Windows in Unite web
2.Do not login with user 2 yet
3.Leave some offline messages to user 2 from user 1.
4.Login with user 2 and make sure you are under Activity List

Issue: After the second user logs in, there is no notification message that informs the user that there are some unread messages.
This should show up on the chat entry in Activity List.
Currently, you see the message user 1 sent (the order in which they show up is wrong UW-363), but you will not know they are new because of the missing notification.

Reproduces on all platforms
FixUnite Web
UW-367Mute microphone and mute speaker (conference controls) require multiple taps/clicksFixed an issue with Conference room controls that required the user to click multiple times to mute microphone or mute speaker.

Reproduce Issue:
1. Using a conference room owner, login into Uunite web
2. Join the conference with the owner, via his phone by dialing the conf room number (you can't drag and drop contacts into conference in android) - linked Jira, and even if you could, you would not be able to answer the incoming invite call - linked Jira
3. From Unite Web, switch to conference bridge
The conf participants will show up here. You have the ability to control microphone, speaker, end call.
4. Tap the microphone icon to enable mute, and then tap it again to unmute.

Issue: while you can tap to mute, it takes 2-3-4 taps to unmute, and then 2-3 more to mute again if you want to.

The same is valid for the mute speaker button.
End call button works fine.

Workaround is to wait 5-10 seconds between each mute/unmute - if that's a workaround

Reproduces on all platforms
FixUnite Web
UW-379Users can't see defined group speed dialsFixed an issue where even though the checkbox is set, the user will not see which speed dials are defined for this group.

Steps to reproduce:
1. Define a user group and add a speed dial to this user group
2. Using a user which is part of this group, login into UW and go under Settings->Speed dials
3. Make sure the Only use group speed dials is checked

If the user goes to old style portal they will see them.
FixUnite Web
UW-383Call history entries time differenceFixed an issue with the UW call history.

Issue Description:
System has current time 14:00.
Users phone has current time 14:00.
User calls some other user and checks the call history in Unite Web.

Issue:
Call entry shows time: 12:00 - 2-hour difference.

Reviewing the System CDR entries, the call entry shows time 14:00
Reviewing the old user portal the call entry shows time 14:00
FixUnite Web
UW-384Disable Dial Pad and Search iconsAn administrator would like to make the Dial Pad and Search icons unavailable to a user or a group of users.

These options would be configurable in the Uniteme Administration GUI in Users -> Users -> “username” or Users -> User Groups -> “user group name”. In the left side menu, there would be a new menu item called Unite Web.

In the Unite Web configuration page there would be the following configuration options for this feature:
Enable Dial Pad Icon Type: Checkbox Default: Enabled
Enable Search Icon Type: Checkbox Default: Enabled
EnhancementUnite Web
UW-385Disable Contact Click to Call and ChatAn administrator would like to disable the ability of a user to use click to call on a contact and also disable the ability of a user to use click to chat.

In the Contacts menu, if a user clicks on a contact’s avatar, information is displayed about that user.

After clicking on the Avatar, information about the contact is displayed.

The first part of this feature request is to be able to disable the click to call capability. The button for this is highlighted. The information should remain (username and extension) but should not allow for click to call.

The second part of this feature request is to be able to disable the click to chat functionality. Click to chat works when a user clicks on the username in the contacts list.

Clicking on the name would normally display a chat area on the right frame.

In the Unite Web configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Contact Click to Call Type: Checkbox Default: Enabled
Enable Contact Click to Chat Type: Checkbox Default: Enabled
EnhancementUnite Web
UW-386Disable Click to Call from Conf BridgeAn Administrator would like to be able to disable the click to call feature from the user’s conference bridge management screen.

Clicking on the highlighted button would normally bring up a dial box where a user can enter an extension or phone number to dial.

In the Unite Web configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Conference Bridge Click to Call Type: Checkbox Default: Enabled
EnhancementUnite Web
UW-387Disable Unite Web menu itemsAn Administrator would like to be able to disable specific menu items in Unite Web and Unite Web Lite.

The menu list is accessed by clicking on the menu button in the upper left.

The administrator would like to be able to control which menu items a user or user group has access to. The administrator would also like to control what settings a user can change.

In the Unite Web configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Activity List Type: Checkbox Default: Enabled
Enable Contacts Type: Checkbox Default: Enabled
Enable Group Chats Type: Checkbox Default: Enabled
Enable Conference Bridge Type: Checkbox Default: Enabled
Enable Voicemails Type: Checkbox Default: Enabled
Enable My Profile Type: Checkbox Default: Enabled
Enable Call History Type: Checkbox Default: Enabled
Enable Settings Type: Checkbox Default: Enabled
Enable Settings Personal Attendant Type: Checkbox Default: Enabled
Enable Settings Call Forwarding Type: Checkbox Default: Enabled
Enable Settings Speed Dials Type: Checkbox Default: Enabled
Enable Settings User Settings Type: Checkbox Default: Enabled
Enable Settings User Settings Change Password Type: Checkbox Default: Enabled
Enable Settings User Settings Voicemail PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Announcement Type: Checkbox Default: Enabled
Enable Settings User Settings eMail Type: Checkbox Default: Enabled
Enable Settings User Settings Attach audio Type: Checkbox Default: Enabled
Enable Settings User Settings Alternate eMail Type: Checkbox Default: Enabled
Enable Settings User Settings Alternate Attach audio Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Room Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Enabled Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Name Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Moderator PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Participant PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Max. members Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Quickstart Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Auto-record Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Moderated Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Public Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Audio source Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Personal MoH Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Files Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Audio file Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Conference enter Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Conference exit Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Voicemail begin Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Voicemail end Type: Checkbox Default: Enabled
Enable Settings Sound Notifications Type: Checkbox Default: Enabled

See Feature 4 in https://docs.google.com/document/d/1wMp1RyFTJiKRNyWWIse1eP_216VaSjwVpBJHso1TPXI/edit#
Is this document something that users can see? If not, we shouldn't reference it here but rather somehow port the content of it to the release notes.
EnhancementUnite Web
UW-388Add conf bridge welcome tones options + ability to disable/enable them as userThe ability to enable or disable entry/exit tones and also for voice announce was added in 16.12.

Add ability for a user to enable/disable conf bridge welcome tones from Unite Lite and Web.

Add ability for a user to enable/disable user announce on entry/exit.
EnhancementUnite Web
UW-391Make Unite Web Tab Blink when there's a new MessageA user would like to see the Unite Web tab blink when there's a new message.EnhancementUnite Web
UW-394There are two download icons in latest ChromeWith the latest version of Chrome, Unite Web users have to download icons for each Voicemail on the Voicemail pageFixUnite Web

 

Issues Sorted by Keyword

 JIRA nameRN ContentEnhancement/Fix/Known IssueKey words
UC-4349Separate backup tar & compressionAn administrator would like to speed up the backup process.

For backup scripts, separately specify the tar and the gzip and set the compression level to 1.

first: tar -cvf
then: gzip -1

This will increase speed but also increase the backup size (but probably not significantly. More concerned with the backup window on larger systems than backup size on smaller systems.
EnhancementBackup
SIPX-526Missing DNS record for Proxy-Forwarding to RegistrarFix for DNS records when a server did not have Registrar enabled.

To reproduce the issue:
Create a cluster with 3 Servers in the following configuration:
PBX01 with Proxy and Registrar
PBX02 with Proxy and Registrar
PBX03 with Proxy

With this setup, every message, sent to the PBX03-Proxy won't get to one of the existing Registrars. Results in lost calls

Config on PBX03-Proxy routes to rr.pbx03.voip.domain.de
This name is not generated in DNS

Current DNS config is:

_sip._tcp.rr IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx01 IN SRV 10 10 5070 pbx01
_sip._tcp.rr.pbx01 IN SRV 30 10 5070 pbx02
_sip._tcp.rr IN SRV 30 10 5070 pbx02
_sip._tcp.rr.pbx02 IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx02 IN SRV 10 10 5070 pbx02

DNS should be configured like this to make it work:

_sip._tcp.rr IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx01 IN SRV 10 10 5070 pbx01
_sip._tcp.rr.pbx01 IN SRV 30 10 5070 pbx02
_sip._tcp.rr IN SRV 30 10 5070 pbx02
_sip._tcp.rr.pbx02 IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx02 IN SRV 10 10 5070 pbx02

_sip._tcp.rr.pbx03 IN SRV 30 10 5070 pbx01
_sip._tcp.rr.pbx03 IN SRV 30 10 5070 pbx02
FixDNS
SIPX-577DNS NAPTR prefer TCPEnhancement to adjust the DNS NAPTR records to have client prefer TCP vs. UDP since TCP is the prefered protocol for SipXcom/UniteMe

Current DNS NAPTR configures them to be equal

voip.domain.de. IN NAPTR 2 0 "s" "SIP+D2U" "" _sip._udp
voip.domain.de. IN NAPTR 2 0 "s" "SIP+D2T" "" _sip._tcp

This should be changed to the following so if some devices use auto configuration, TCP will be chosen.

voip.domain.de. IN NAPTR 2 0 "s" "SIP+D2U" "" _sip._udp
voip.domain.de. IN NAPTR 1 0 "s" "SIP+D2T" "" _sip._tcp
FixDNS
SIPX-572Jitsi Provisioning DNDAn enhancement to Jitsi provisioning with the following parameter:

net.java.sip.communicator.impl.protocol.RejectIncomingCallsWhenDnD

For sipXcom/Uniteme it is necessary to set this parameter has to be set to "true", because the Proxy/Registrar could not handle the DND itself.
EnhancementJitsi
UC-4363Jitsi provisioning automatic display nameAn administrator would like such that if nothing is entered for the Display name, that the provisioning fills this with name and surname stored in the user profile by itself (same behavior as Polycom Plugin).

The current Version of Jitsi Provisioning can configure the line display name (Lines > SIP > Displayname), it's just not utilizing the name and surname from the user profile.
EnhancementJitsi
UC-4376Disable MWI subscription if Voicemail permissions are disabledFixed an issue where Polycom phones are trying aggressively to subscribe for Event: message-summary.
To those Subscribes proxy returns 403 but server is flooded by SUBSCRIBES by the users that have Voicemail permissions are disabled.
FixPolycom
UC-4311Polycom SoundPoint IP 650 and 560 background images are not workingFixed an issue with background images on Polycom SPIP 650 and 560 phones.FixPolycom sipXconfig
UC-2879Allow IVR to set a Reach Agent Specific Skill

Enhancement to allow the IVR to set an agent specific skill for a call.

Currently, an IVR script can set skills required for a Reach call prior to sending the call along to Reach. This is done by setting the "skills" variable via a command such as:
skills = "skill1,skill2,skill3";
session:setVariable("skills", skills);

Each of the skills "skill1", "skill2" and "skill3" would then be built in the Reach configuration and those calls coming from the IVR would require those skills.

Note that the default weight of this call should be higher than other calls by default such that the agent isn't offered a regular queue call prior to being offered this agent specific call. This would be similar to how a transfer to agent works (internally treated with higher priority).

It is desirable to allow the IVR to set an agent specific skill for a call as well. This should be done in the same way as setting other skills.

Use Case:
1. caller calls into IVR
2. IVR collects a ticket number
3. IVR dips a back end system to see what agent is assigned to this ticket
4. IVR builds skill list with base skills such as "ServiceDesk", "Tech", etc AND the agent specific skill that is assigned to the ticket.

EnhancementReachme
UC-3573"This week" queue statistics seen by Supervisor are cleared on TuesdayFix for "This week" queue statistics which were being cleared on Tuesdays.

To reproduce:
1. On Monday - Place a few calls to make sure the Queue statistics get populated.
The Queue statistics have been updated ok for all of the options (Last 15 Minutes, Today .. This Month etc).
2.Next day, Tuesday, check the statistics

Issue: Statistics showed that "This Week" statistics was 0. (as if yesterday was Sunday... and the week ended)

Expected: Statistics to show a number of calls placed Monday.

Place a few more calls and "This week" will get populated with statistics based on calls that were just placed.
FixReachme
UC-4228Add agent specific skill to remove skill recipe actionEnhancement to add a "special entry" in the list of skills to possibly remove in the remove skills recipe action.

Currently, when building a remove skills recipe action in the Reach admin, the skills listed in the action box that you can remove include the following:
All skills that the admin has added to the system
Magic skills of:
Node
Queue
All
Client

This proposed feature is to allow the admin to remove any agent-specific skill.

Admin UI should display a new value under the Magic subsection. This new value will be "Agent".

If selected, when the recipe step is executed based on the criteria of the recipe step being matched, Reach should remove any/all agent specific skills from the call.

This relates to calls that have been transferred to a specific user. In such a case, there is an agent specific skill added to the call when it is transferred.

More importantly, it relates to UC-2879 which adds the ability for an IVR script to add an agent specific skill. Once that is added, then we need to have the ability to remove this agent specific skill from the call based on a recipe step.

The use case is as follows:
Call comes into IVR
IVR interacts with the caller and with a back end ticket system
IVR finds that there is an open ticket in the ticketing system
IVR finds the "ticket owner" from the back office system (e.g agent 2222 is the owner of the ticket)
IVR sets brand, queue and skills on call; skills include agent specific skill of 2222
IVR sends call to reach queue

Now, reach will only deliver the call to the specific agent 2222 since that agent specific skill 2222 is only assigned to agent 2222
At this point, you would then want to be able to remove this skill so that the call could go to other agents (e.g. agent 2222 is not logged in).
EnhancementReachme
UC-4316Improve Reachme agent reconnect in case of network loss - leave call upEnhancement to the Reachme agent/supervisor portal for reconnecting in the event of network connectivity loss.

In the current situation if the agent loses network connection to the Reach server, the reconnection mechanism will detect this connection loss and try to recover from it in two ways:
1. If the drop is less than 17 seconds in length the agent does not get released or logged out and his call only loses voice path for these few seconds, call not getting dropped.
2. If the drop is bigger, then the agent is prompted that connection to the server is lost, the agent is released, and the call is dropped in approx 40 seconds.

However, it seems our customers would like to have this feature somehow improved.
Customers are saying: "The Reach web client has a very tight connection with a very low tolerance for connections blips, making it uniquely fragile in situations like this, causing not only the calls to drop or re-queue, but all the agents to release."

Some tests we've executed:
https://docs.google.com/spreadsheets/d/1HHdFH6FD6e86qcErwG_mOnf09XHGn2020KjTdef0Cwg/edit#gid=0

CURRENT PLAN FOR FIX IS AS FOLLOWS:
1. If the WebSocket connection is detected from UI side, attempt to reconnect as usual. This will include the new fix for the issue where a reconnect would not be successful unless the URL was an exact match to the standard reach URL.
2. If the ReachMe server side recognizes a WebSocket disconnect, it will
- leave the call up
- not attempt to use the socket connection any further
- show the agent in the agent manager as still being on a call
- record complete CDR for the call as it does for any other normal call
- Then, at the end of the call, kill off the agent connection.
3. If the UI socket connection is lost and the agent still has an active call, the agent or the system may reconnect. If they do while the call is still active, they will receive a warning that reconnecting the UI session will end the active call. If they proceed, the call will be hung up (just like any other agent initiated hang up) and the UI session will be started new.
EnhancementReachme
UC-4330When agent connects via IP address instead of FQDN, reconnection mechanism does not workFixed an issue with the Reachme portal reconnection mechanism where it would appear to work if you used an IP instead of the server name, but the server connection would, in fact, be disconnected.

To reproduce the issue:
If you insert the server's IP address e.g. 10.5.0.210/reach to get to the dashboard, then you would normally get automatically redirected to the "server_fqdn"/reach.
But then if you insist and edit that address, and replace the FQDN with the IP address, then, in this case, it allows you to use it, and it does not redirect you any more to server's FQDN.

In this case, after you login and a network disconnection occurs (larger than 15s), the dashboard will "appear" to work but in fact, it won't.
If you click the Release button, it gets grayed out.
If you place calls in the queue, the agent is not alerted.

The agent must refresh the page to make it work again.
FixReachme
UC-4354Allow Reach Admin to Define Call Recovery MechanismFixed an issue where users of Unite Web were not getting unread messages they may have been sent while they were offline.

Reproduce Issue:
Two users must exist on the system.
1. Login with user 1 on Firefox/Chrome Windows in Unite web
2. Do not login with user 2 (yet)
3. Leave some offline messages to user 2 from user 1.
4. Login with user 2 and make sure you are under Activity List

Issue: After the second user logs in, there is no notification message that informs the user that are some unread messages.
This should show up on the chat entry in Activity List.
Currently, you see the message user 1 sent (the order in which they show up is wrong UW-363), but you will not know they are new because of the missing notification.

Reproduces on all platforms
EnhancementReachme
UC-4355Change Reach call recording archival to be more intuitiveThere is a current setting in the Client record in the reach configuration called 'Days to Retain Recordings in Archive'. This value CURRENTLY represents the total number of days that a recording has been in the system before it should be removed from the archival location.

An example of how it was working:
A call comes into the system on 2/1 and is set to be recorded and kept in the system locally for 5 days and then be archived. The 'Days to Retain Recordings in Archive' is set to 10. The recording will be made and placed into the local system for 5 days. It will then me moved to the archive and reside there for an additional 5 days (total of 10).

This Jira is a request to change that the archiving back end operates in the same manner as the GUI says it will.

An example of how it should work after this fix:
A call comes into the system on 2/1 and is set to be recorded and kept in the system locally for 5 days and then be archived. The 'Days to Retain Recordings in Archive' is set to 10. The recording will be made and placed into the local system for 5 days. It will then me moved to the archive and reside there for an additional 10 days (total of 15).
FixReachme
UC-4389Agent State History/Agent Unanswered Call Details missing when upgrading from 16.04.stage/16.08.stage to 16.12.stageFixed an issue with Agent State History and Agent Unanswered Call Details missing when upgrading from 16.04.stage/16.08.stage to 16.12.stage.
Also, the Agent Group prod & Agent Productivity by Group fail to open.
FixReachme
UC-4141Ensure Reach Reports are backed up with the backup process - standard and customEnhancement to ensure that all standard deployed Reports, as well as any custom Reports that may have been deployed, are backed up as part of the system backup.

Obviously, underlying data used by the Reports should also be backed up.

Restoring this backup should result in getting all of the standard Reports, still having any custom Reports that were previously created and data required to run the Reports.
EnhancementReachme Backup
UC-4061Add Agent States Reach ReportEnhancement to add a Reachme report with the following details.

Inputs:
Date/Time, skills list (multi-select; default that to all skills)

Outputs:
Agent name, Login, Current state during the input date/time

Additional Output
A separate table formatted output with a summary of the data shown in the details section defined above. This summary section will have columns headers of ...
Available, Ringing, On Cal, Wrap Up, Outbound, Other, Release1, Release 2 ... Release "n" (an entry for each release reason built in the system)
Data under these headers will be a total count of agents that were in that state
EnhancementReachme Reports
UC-4146Update Agent Group Activity Reach Report to have Outbound Sent ColumnEnhancement to the Agent Group Activity report to add Outbound Sent column.

Update this report as follows:

Under the Outbound subsection;
Change total outbound column name to Started (total CDRs with direction outbound)
Add column called Sent under the Outbound activity area (total CDRs with direction outbound and disposition agent initiated)
EnhancementReachme Reports
UC-4201Add a Voicemail Overview ReportFixed an issue with the Reachme portal reconnection mechanism where it would appear to work if you used an IP instead of the server name, but the server connection would, in fact, be disconnected.

To reproduce the issue:
If you insert the server IP address e.g. 10.5.0.210/reach to get to the dashboard, then you would normally get automatically redirected to the "server_fqdn"/reach.
But then if you insist and edit that address, and replace the FQDN with the IP address, then, in this case, it allows you to use it and it does not redirect you any more to server's FQDN.

In this case, after you login and a network disconnection occurs (larger than 15s), the dashboard will "appear" to work but in fact, it won't.
If you click the Release button, it gets grayed out.
If you place calls in the queue, the agent is not alerted.

The agent must refresh the page to make it work again.
EnhancementReachme Reports
UC-4202Add a Voicemail Detail ReportEnhancement to add a new report that offers detailed information on Voicemail activity. This is the same as UC-4201 except with an additional input parameter of interval and the output is grouped by the interval input. Interval should default to 60 minutes.EnhancementReachme Reports
UC-4207Agent State History report shows "null" when selected agent has no activityFix to solve the case where the report shows "null" when a selected agent has no activity.

To reproduce
1. Create a new agent under some agent group
Don't do anything with this agent
2. Login with a supervisor, go to Reports, select Agent State History report and the correct timeframe.
Select the agent group, and then choose the newly added agent
3. Generate report
Issue: report shows "null"
FixReachme Reports
UC-4284Add magic skills to Agent Availability reportEnhancement to the Agent Availability report to also show the Magic Skills.EnhancementReachme Reports
UC-4343Reachme Reports section does not display the full set of input controls - missing scrollbarFixed an issue that was presented when a report had multiple input controls. When this was the case, the display window was not large enough. The solution is to implement a scrollbar for the window.FixReachme Reports
UC-3448SAA does not apply some configuration file parametersFix for 2 configuration file parameters which were not applied to SAA service, so SAA would always work with its defaults for this params:
server-min-expires (300 secs)
server-default-expires (3600 secs)

Steps to reproduce:
1. Change the values of the 2 params in the openuc-saa.ini file (i.e. change them to 60)
2. Start SAA with INFO log level
3. Grep after the "expire" message in openuc-saa's log file:
qq
"2014-12-12T15:53:48.977944Z":50:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSClientSubscriptionHandler::SSSClientSubscriptionHandler::0x7fffc3bbfe10 Client subscription default expire seconds = 3600"
"2014-12-12T15:53:48.978167Z":56:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerSubscriptionHandler::SSSServerSubscriptionHandler::0x7fffc3bbfee0 Server subscription min expires = 60"
"2014-12-12T15:53:48.978197Z":57:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerSubscriptionHandler::SSSServerSubscriptionHandler::0x7fffc3bbfee0 Server subscription default expires = 60"
"2014-12-12T15:53:48.978219Z":58:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerSubscriptionHandler::SSSServerSubscriptionHandler::0x7fffc3bbfee0 Server subscription max expires = 86400"
"2014-12-12T15:53:48.978353Z":64:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerBase::readConfigOptions::0x7fffc3bbf980 min expires = 300"
"2014-12-12T15:53:48.978383Z":65:SIP:INFO:ioancea.fedora19::7f455a84c900:openuc-saa:"SSSServerBase::readConfigOptions::0x7fffc3bbf980 default expires = 3600"
qq
As can be seen, the SSSClientSubscriptionHandler::SSSClientSubscriptionHandler() and SSSServerBase::readConfigOptions() functions are using the default values: 300 and 3600, respectively.

Also, the values can also be checked at runtime: client subscriptions are always created using the default Expire (i.e. 3600) and the minimum expiration (which can be seen when SAA refreshes the SUBSCRIPTION during a call) is always the default one (i.e. 300)
FixSAA
SIPX-569Syslog only receiving on UDP 514In older sipXcom releases, the default Syslog transport setting for Polycom phone groups was UDP. In release 16.02, the default setting syslog transport setting is TCP, which triggers the phones to send log file information to Sipxcom using TCP transport port 1468.

This is a fix to set syslog transport back to UDP which is preferred over TCP to preserve TCP sockets.
FixsipXconfig
SIPX-581Update Admin GUI w/Blog Posts and LinksThis is an enhancement to the Admin GUI to receive blog post updates from sipxcom.org website in the Admin GUI as well as a link to important sipxcom.org URLs.

Add RSS feed capabilities to Admin GUI. The feed should come from http://sipxcom.org/feed/

Add the important links on the right as shown in Mockup:

Downloads (ISO's & RPM's): sipXcom ISO Images and RPM Repositories
(link the text after the ':' to http://wiki.sipxcom.org/display/sipXcom/sipXcom+ISO+Images+and+RPM+Repositories)
JIRA Issue Tracker: http://jira.sipxcom.org
sipXcom on Github: https://github.com/sipXcom
sipXcom User's Mailing List: https://groups.google.com/d/forum/sipxcom-users
sipXcom Developers Mailing List: https://groups.google.com/d/forum/sipxcom-dev
Paid Version & Optional Features: https://www.ezuce.com

At top of the page darken icons to 50% gray. Add Paid Support, change 'Help' to 'Docs'.

Paid Support should link to: http://ezuce.com/products-solutions/procare-sipxcom-support
EnhancementsipXconfig
UC-3683REST API to create/modify a user/user group and set propertiesEnhancement to allow for user management through a rest API.

Currently, we have SOAP support for user creation, but there is no support for user settings (like user called id for example)

Use the new REST API engine based on Apache CXF to create such API. The settings management is generally handled by current existing REST support
EnhancementsipXconfig
UC-4296Web UI Search searches Voicemail PIN TokenFixed an issue where Voicemail PIN was included as a searchable field when searching for an extension in the system.

To reproduce the issue:
Example: You search for "96"

The search will show you every user starting with 96 and user with a voicemail pin token that starts with 96
FixsipXconfig
UC-4297Increase elasticsearch user limitsFix for the Elasticsearch user file limits which are set too low for larger systems.

We should have an entry in /etc/security/limits.d/ for user elasticsearch to increase currently the number of open files and processes that user can spawn
per this description: https://www.elastic.co/guide/en/elasticsearch/guide/current/_file_descriptors_and_mmap.html
FixsipXconfig
SIPX-587Rotate proxy_stats.json file - proxy statsThis is an enhancement in support of the proxy statistics enhancement. The file where proxy stats are collected should be included in the Logrotate mechanism.EnhancementsipXconfig sipxproxy
SIPX-578Stats collecting submodule for ProxyEnhancement to sipXproxy that is required to implement internal metrics collecting submodule for proxy and output collected metrics into a file with a pre-configured interval.

In the first approach, Proxy shall have statistics file with name=value content (like "ProxyQueueSize=123"). Single line for a single metric.
The file should be updated every StatsUpdateInterval seconds configuration parameter. The default value will be 15 seconds. In the future, if we will need we can add this to WebUI configuration.
EnhancementsipXproxy
UC-4315SSS crash on empty "Refer-To" fieldFixed an issue with SSS where it would crash if it got messages with empty Refer-To field.FixSSS
UC-4290Enhancement request for Unite Web CDR durationAn end user would like to see a call Duration column (or, alternatively, the End Date/Time) in the end user Call Detail History in Unite Web.EnhancementUnite Web
UC-4324Unite Web call history improvementsEnhanced the Unite Web Call History.

These are some of the changes that need to be implemented:
1.show timezone drop down as in old style portal, but defaulting to show the timezone of the user's PC (old style portal relies on what is set under User->Time Zone, but we don't need that)
2.Show call history entries by default, based on the default selection.
As soon as you go on the call history page, without hitting Apply.
3.Reverse Apply button location with To/from box
4.When using To/From box, results should show up if you both hit Enter or click Apply
5.Fix time format to 00:00:00 in Start/Stop columns instead of 00:0:00
6.Sorting of columns in results, if it's easy
EnhancementUnite Web
UW-355On Safari and IOS, the scrolling is not working properlyFixed an issue under the Profile tab that was reproduced after the user tried to edit some fields and then scroll down and save it. After saving, when the user would scroll up to the beginning the user would notice that the scroll is working very hard and sometimes is not working at all (because the user can accidentally scroll from the margin and as result entire web page will be scrolled).

If the user scrolls up or down from the middle of the screen then it will work ok without problems.
FixUnite Web
UW-362No new message notification in Activity List for unread messages received while offlineFixed an issue where users of Unite Web were not getting unread messages they may have been sent while they were offline.

Reproduce Issue:
Two users must exist on the system.
1.Login with user 1 on Firefox/Chrome Windows in Unite web
2.Do not login with user 2 yet
3.Leave some offline messages to user 2 from user 1.
4.Login with user 2 and make sure you are under Activity List

Issue: After the second user logs in, there is no notification message that informs the user that there are some unread messages.
This should show up on the chat entry in Activity List.
Currently, you see the message user 1 sent (the order in which they show up is wrong UW-363), but you will not know they are new because of the missing notification.

Reproduces on all platforms
FixUnite Web
UW-367Mute microphone and mute speaker (conference controls) require multiple taps/clicksFixed an issue with Conference room controls that required the user to click multiple times to mute microphone or mute speaker.

Reproduce Issue:
1. Using a conference room owner, login into Unite Web.
2. Join the conference with the owner, via his phone by dialing the conf room number (you can't drag and drop contacts into conference in android) - linked Jira, and even if you could, you would not be able to answer the incoming invite call - linked Jira
3. From Unite Web, switch to conference bridge
The conf participants will show up here. You have the ability to control microphone, speaker, end call.
4. Tap the microphone icon to enable mute, and then tap it again to unmute.

Issue: while you can tap to mute, it takes 2-3-4 taps to unmute, and then 2-3 more to mute again if you want to.

The same is valid for the mute speaker button.
End call button works fine.

Workaround is to wait 5-10 seconds between each mute/unmute - if that's a workaround

Reproduces on all platforms
FixUnite Web
UW-379Users can't see defined group speed dialsFixed an issue where even though the checkbox is set, the user will not see which speed dials are defined for this group.

Steps to reproduce:
1. Define a user group and add a speed dial to this user group
2. Using a user which is part of this group, login into UW and go under Settings->Speed dials
3. Make sure the Only use group speed dials is checked

If the user goes to old style portal they will see them.
FixUnite Web
UW-383Call history entries time differenceFixed an issue with the UW call history.

Issue Description:
System has current time 14:00.
Users phone has current time 14:00.
User calls some other user and checks the call history in Unite Web.

Issue:
Call entry shows time: 12:00 - 2-hour difference.

Reviewing the System CDR entries, the call entry shows time 14:00
Reviewing the old user portal the call entry shows time 14:00
FixUnite Web
UW-384Disable Dial Pad and Search iconsAn administrator would like to make the Dial Pad and Search icons unavailable to a user or a group of users.

These options would be configurable in the Uniteme Administration GUI in Users -> Users -> “username” or Users -> User Groups -> “user group name”. In the left side menu, there would be a new menu item called Unite Web.

In the Unite Web configuration page there would be the following configuration options for this feature:
Enable Dial Pad Icon Type: Checkbox Default: Enabled
Enable Search Icon Type: Checkbox Default: Enabled
EnhancementUnite Web
UW-385Disable Contact Click to Call and ChatAn administrator would like to disable the ability of a user to use click to call on a contact and also disable the ability of a user to use click to chat.

In the Contacts menu, if a user clicks on a contact’s avatar, information is displayed about that user.

After clicking on the Avatar, information about the contact is displayed.

The first part of this feature request is to be able to disable the click to call capability. The button for this is highlighted. The information should remain (username and extension) but should not allow for click to call.

The second part of this feature request is to be able to disable the click to chat functionality. Click to chat works when a user clicks on the username in the contacts list.

Clicking on the name would normally display a chat area on the right frame.

In the Unite Web configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Contact Click to Call Type: Checkbox Default: Enabled
Enable Contact Click to Chat Type: Checkbox Default: Enabled
EnhancementUnite Web
UW-386Disable Click to Call from Conf BridgeAn Administrator would like to be able to disable the click to call feature from the user’s conference bridge management screen.

Clicking on the highlighted button would normally bring up a dial box where a user can enter an extension or phone number to dial.

In the Unite Web configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Conference Bridge Click to Call Type: Checkbox Default: Enabled
EnhancementUnite Web
UW-387Disable Unite Web menu itemsAn Administrator would like to be able to disable specific menu items in Unite Web and Unite Web Lite.

The menu list is accessed by clicking on the menu button in the upper left.

The administrator would like to be able to control which menu items a user or user group has access to. The administrator would also like to control what settings a user can change.

In the Unite Web configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Activity List Type: Checkbox Default: Enabled
Enable Contacts Type: Checkbox Default: Enabled
Enable Group Chats Type: Checkbox Default: Enabled
Enable Conference Bridge Type: Checkbox Default: Enabled
Enable Voicemails Type: Checkbox Default: Enabled
Enable My Profile Type: Checkbox Default: Enabled
Enable Call History Type: Checkbox Default: Enabled
Enable Settings Type: Checkbox Default: Enabled
Enable Settings Personal Attendant Type: Checkbox Default: Enabled
Enable Settings Call Forwarding Type: Checkbox Default: Enabled
Enable Settings Speed Dials Type: Checkbox Default: Enabled
Enable Settings User Settings Type: Checkbox Default: Enabled
Enable Settings User Settings Change Password Type: Checkbox Default: Enabled
Enable Settings User Settings Voicemail PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Announcement Type: Checkbox Default: Enabled
Enable Settings User Settings eMail Type: Checkbox Default: Enabled
Enable Settings User Settings Attach audio Type: Checkbox Default: Enabled
Enable Settings User Settings Alternate eMail Type: Checkbox Default: Enabled
Enable Settings User Settings Alternate Attach audio Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Room Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Enabled Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Name Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Moderator PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Participant PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Max. members Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Quickstart Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Auto-record Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Moderated Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Public Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Audio source Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Personal MoH Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Files Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Audio file Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Conference enter Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Conference exit Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Voicemail begin Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Voicemail end Type: Checkbox Default: Enabled
Enable Settings Sound Notifications Type: Checkbox Default: Enabled

See Feature 4 in https://docs.google.com/document/d/1wMp1RyFTJiKRNyWWIse1eP_216VaSjwVpBJHso1TPXI/edit#
Is this document something that users can see? If not, we shouldn't reference it here but rather somehow port the content of it to the release notes.
EnhancementUnite Web
UW-388Add conf bridge welcome tones options + ability to disable/enable them as userThe ability to enable or disable entry/exit tones and also for voice announce was added in 16.12.

Add ability for a user to enable/disable conf bridge welcome tones from Unite Lite and Web.

Add ability for a user to enable/disable user announce on entry/exit.
EnhancementUnite Web
UW-391Make Unite Web Tab Blink when there's a new MessageA user would like to see the Unite Web tab blink when there's a new message.EnhancementUnite Web
UW-394There are two download icons in latest ChromeWith the latest version of Chrome, Unite Web users have to download icons for each Voicemail on the Voicemail pageFixUnite Web
UC-4313Conference settings REST API to include new parametersThis is a Rest API enhancement to support Unite Web and Unite Lite enhancements.

We need to add new parameters:
Play Entry Tone Type: Checkbox
Play Exit Tone Type: Checkbox
Play Voice Announce Entry Type: Checkbox Default: Enabled
Play Voice Announce Exit Type: Checkbox Default: Enabled

to existing REST API
method: 'GET' and 'PUT'
/my/conferences/"conference name"
EnhancementUnite Web Conferencing sipXconfig
UC-4307Add new settings for User Portal configurationThis is an enhancement in support of Unite Web work to allow an Administrator to be able to control the enabling and disabling of features of the User Portal by User and by User Group.

This work is in support of the 4 new features.

Feature 1 - Disable Dial Pad and Search icons
These options would be configurable in the Uniteme Administration GUI in Users -> Users -> “username” or Users -> User Groups -> “user group name”. In the left side menu, there would be a new menu item called User Portal.

In the User Portal configuration page there would be the following configuration options for this feature:
Enable Dial Pad Icon Type: Checkbox Default: Enabled
Enable Search Icon Type: Checkbox Default: Enabled

Feature 2 - Disable Contact Click to Call and Chat
In the User Portal configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Contact Click to Call Type: Checkbox Default: Enabled
Enable Contact Click to Chat Type: Checkbox Default: Enabled

Feature 3 - Disable Click to Call from Conf Bridge
In the User Portal configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Conference Bridge Click to Call Type: Checkbox Default: Enabled

Feature 4 - Disable Unite Web menu items
In the Unite Web configuration page (in Users -> Users -> “username” and Users -> User Group -> “user group name”) there would be the following configuration options for this feature:
Enable Activity List Type: Checkbox Default: Enabled
Enable Contacts Type: Checkbox Default: Enabled
Enable Group Chats Type: Checkbox Default: Enabled
Enable Conference Bridge Type: Checkbox Default: Enabled
Enable Voicemails Type: Checkbox Default: Enabled
Enable My Profile Type: Checkbox Default: Enabled
Enable Call History Type: Checkbox Default: Enabled
Enable Settings Type: Checkbox Default: Enabled
Enable Settings Personal Attendant Type: Checkbox Default: Enabled
Enable Settings Call Forwarding Type: Checkbox Default: Enabled
Enable Settings Speed Dials Type: Checkbox Default: Enabled
Enable Settings User Settings Type: Checkbox Default: Enabled
Enable Settings User Settings Change Password Type: Checkbox Default: Enabled
Enable Settings User Settings Voicemail PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Announcement Type: Checkbox Default: Enabled
Enable Settings User Settings eMail Type: Checkbox Default: Enabled
Enable Settings User Settings Attach audio Type: Checkbox Default: Enabled
Enable Settings User Settings Alternate eMail Type: Checkbox Default: Enabled
Enable Settings User Settings Alternate Attach audio Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Room Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Enabled Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Name Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Moderator PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Participant PIN Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Max. members Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Quickstart Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Auto-record Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Moderated Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Public Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Entry Tone Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Exit Tone Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Voice Announce Entry Type: Checkbox Default: Enabled
Enable Settings User Settings Conference Bridge Voice Announce Exit Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Audio source Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Personal MoH Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Files Type: Checkbox Default: Enabled
Enable Settings User Settings MoH Audio file Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Conference enter Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Conference exit Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Voicemail begin Type: Checkbox Default: Enabled
Enable Settings User Settings MyBuddy Voicemail end Type: Checkbox Default: Enabled
Enable Settings Sound Notifications Type: Checkbox Default: Enabled
EnhancementUnite Web sipXconfig
UC-4328REST API to manage user properties for user portalAn Enhancement to support the work for Unite Web work. Provide set of rest apis that a regular user (USER_ROLE) can access, in order to update logged in user properties/settingsEnhancementUnite Web sipXconfig
SIPX-539Yealink Emergency DND FeatureEnhancement to allow provisioning support for Yealink's Emergency DND Feature.

From Yealink Provisioning Guide:

Specify the authorized numbers when DND is enabled.
Parameters:
features.dnd.emergency_enable
features.dnd.emergency_authorized_number
EnhancementYealink
SIPX-540Yealink Call Number FilterEnhancement to allow provisioning support for Yealink's Call Number Filter.

From Yealink Provisioning Guide:

Configure the characters the IP phone filters when dialing.
Parameters:
features.call_num_filter
EnhancementYealink
SIPX-560Alert Info ExternalThis is an improvement of Proxy Plugin to set Alert-Info-Header.

Some phones (e.g. Yealink) bypass the Proxy if the From-Header do not end with @<sipdomain>.

For SIP-Devices that have no ability to add custom headers to a SIP Message (e.g. Patton), it is necessary to scan the From header for a tag (x-sipx-alert-info=external) to set the Alert-Info Header for From-Uris with the SIP Domain inside.
EnhancementYealink
SIPX-565Yealink Provisioning of Voice Quality MonitoringEnhancement to the Yealink provisioning to enable Yealink configuration of RTCP-XR parameters.

Yealink can send a Report to a data collector to get information about the quality of the last call.

Event is: vq-rtcpxr

On activation, the Phone sends a Publish Message with the Report to a service you can configure
EnhancementYealink
SIPX-585Yealink Resource List SubscriptionFix for Yealink Provisioning to check if lines on the phone have BLFs.

Current Issue:
If a user has no BLFs, Yealink phone would start to spam to proxy with resource list subscriptions and if you have enough Yealink to proxy stops working.
FixYealink
SIPX-600Custom settings enhancement for Yealink provisioningEnhancement to add the ability to add custom settings to Yealink configuration.

This is a good feature with Polycom provisioning which is now available for Yealink provisioning. Settings which are currently missing in GUI could be set via this custom config.
EnhancementYealink

 



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