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  • sipXcom 15.10

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Summary

 

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New Features:

    • Locations Feature to Block Calling between Locations

    • Call Back on Busy now Highly Available

    • Set Polycom Background Images in Admin GUI

    • Logging of Authorization Codes to Call Detail Report

    • Provision Jitsi from Admin GUI

    • Obfuscate CDR Records

Enhancements:

    • New eZuce Logo on Polycom Phones

    • Audit Log Performance Enhancements

    • Configure Outbound Port for Exchange Voicemail

    • Update of Freeswitch to 1.4.20

    • Included VP8 Codec in Media Services

    • Yealink phone enhancements / fixes

    • Polycom phone enhancements / fixes

    • Allow for Polycom firmware up to version 5.4.0

 

Who Should Install?

This release is recommended for all 4.6, 14.XX and 15.XX installations.

New Installs

A new ISO is available for 15.10 at: http://download.sipxcom.org/pub/sipXecs/ISO/

Update

To update please edit your /etc/yum.repos.d/sipxecs.repo file and reference the new download server (download.sipxcom.org).  The repo should look as follows:

[sipXcom]
name=sipXecs software for CentOS $releasever - $basearch
baseurl=http://download.sipxcom.org/pub/sipXecs/15.10/CentOS_$releasever/$basearch
gpgcheck=0

 

To edit this file, login to your sipX server as root and then use either vi or nano (easier).

vi /etc/yum.repos.d/sipxecs.repo

   or

nano /etc/yum.repos.d/sipxecs.repo

 

Once the repo file is modified, run:

yum clean all

yum update


Issues Addressed

 JIRA nameRN ContentEnhancement/Fix/Known IssueKey words
SIPX-314Obfuscate CDRsAdded a new feature to obfuscate (masquarade) parts of dialed phone numbers. Larger companies sometimes have concerns about the data privacy of the cdrs.

Settings can be found in System -> CDR -> Advanced Settings
Enable Masquerading - enable / disable the freature
Minimum length of extension that muse be masqueraded - This value is the minimum length of extension that should be covered with a *
Prefixes excluded from masquerading - Certain numbers prefixed with these values will show the entire dialed number and not be covered with a *
EnhancementCDR
SIPX-84Log Accepted authorization code.Added the ability to log the accepted Authorization Code and store in CDR.EnhancementCDR, AuthCode
SIPX-167Disable ipv6 using cfengine promiseFixed an issue with RPM installations to disable IPV6. This was being done properly for ISO installations.

Ensure the existence of:

net.ipv6.conf.all.disable_ipv6 = 1
net.ipv6.conf.default.disable_ipv6 = 1

is within /etc/sysctl.conf until we have completed ipv6 support.
EnhancementConfig
SIPX-256Make call back on busy HAMake call back on busy a highly available service.EnhancementConfig
SIPX-287Add fields to confirm passwords on Admin Settings and SNMP pagesAdd confirmation fields for default password,default PIN and default psql password on Admin Settings page and confirmation field for community string on SNMP page.EnhancementConfig
UC-3666Feature request - ability to assign port number to exchange voicemail dial planAdded the ability to set a port in the Voicemail dialplan when using Exchange Voicemail Server as an option.

Needed the ability to change the port to 5080 when sending and not use 5060. This eliminated the need to use an SBC to communicate to Office 365 Exchange Unified Messaging
EnhancementConfig
SIPX-226Option to enable Core Dumps for FreeswitchAdded a new option in System -> Media Services -> Advanced Settings to enable core dumps.

in /etc/sysctl.conf add

fs.suid_dumpable=2
kernel.core_pattern=/tmp/core

then

/sbin/sysctl -p /etc/sysctl.conf

verify properly set

[root@openuc ~]# sysctl fs.suid_dumpable
fs.suid_dumpable = 2

In /etc/init.d/sipxfreeswitch add -core switch when starting up, e.g.

DAEMON_START_ARGS="-nc -nonat -core \
-u sipx \
-g sipx \
$DAEMON_ARGS"

then restart freeswitch service as service sipxfreeswitch restart.

Force core dump as:

fs_cli
freeswitch@internal> fsctl crash
Socket interrupted, bye!

Core file generated as

[root@openuc ~]# ls -la /tmp/core.20319
-rw------- 1 sipx sipx 28315648 Jul 21 04:06 /tmp/core.20319

(pattern is core.PID)

To see what did the crash (mind that freeswitch-debuginfo RPM should be installed and same version as freeswitch):

gdb freeswitch /tmp/core.20319

then type bt, you will get something as

(gdb) bt
#0 0x00007fe36c705625 in raise () from /lib64/libc.so.6
#1 0x00007fe36c706e05 in abort () from /lib64/libc.so.6
#2 0x00007fe36ed59c61 in switch_core_session_ctl (cmd=SCSC_CRASH, val=0x0) at src/switch_core.c:2506
#3 0x00007fe35a3d49c5 in ctl_function (cmd=<value optimized out>, session=<value optimized out>, stream=0x7fe34943cae0) at mod_commands.c:2219
#4 0x00007fe36ed873de in switch_api_execute (cmd=0x7fe34c00bd80 "fsctl", arg=0x7fe34c00bd86 "crash", session=0x0, stream=0x7fe34943cae0) at src/switch_loadable_module.c:2438
#5 0x00007fe36ed3153b in switch_console_execute (xcmd=<value optimized out>, rec=0, istream=0x7fe34943cae0) at src/switch_console.c:391
#6 0x00007fe35b999a75 in api_exec (thread=<value optimized out>, obj=0x7fe34943d470) at mod_event_socket.c:1503
#7 0x00007fe35b99b4cd in parse_command (listener=<value optimized out>, event=0x7fe34943d758, reply=0x7fe34943db70 "", reply_len=512) at mod_event_socket.c:2253
#8 0x00007fe35b99d9b7 in listener_run (thread=<value optimized out>, obj=0x7fe36000b208) at mod_event_socket.c:2677
#9 0x00007fe36d0bd9d1 in start_thread () from /lib64/libpthread.so.0
#10 0x00007fe36c7bb8fd in clone () from /lib64/libc.so.6

or

bt full
thread apply all bt
thread apply all bt full
EnhancementConfig, Media
SIPX-239Upgrade FS to latest versionUpgrade FS to latest version (1.4.20)EnhancementConfig, Media
SIPX-326Add vp8 codec config for media servicesEnabled the vp8 codec in media servicesEnhancementConfig, Media
SIPX-228Provision JitsiBe able to provision Jitsi as a phone and XMPP client from within sipXcom.

In phone and phone groups be able to create a Jitsi phone and associate a user with it.

Enable the ability to add XMPP capability if the user has sipXcom XMPP configured.

Configuration should work much like existing Bria configuration capability.

https://jitsi.org/Documentation/ConfigureJitsiWithProvisioning
EnhancementJitsi
SIPX-267Change System -> Branches to System -> LocationsAs part of the new Locations feature, change the Branches menu item to be Locations.EnhancementLocations
SIPX-269In Locations (Branches) page add Calling RestrictionsAdded the ability to restrict calling based on a user's defined Location.

The Administrator would like to be able to configure which other Locations users in a particular Location are allowed to call.

The Administrator would like to be able to control what features users from other Locations are allowed to access for a particular Location (i.e., Auto Attendants, Hunt Groups, Intercom, Paging, Conference Bridge, Call Queue, Park Orbits, Authorization Codes).
EnhancementLocations
SIPX-285Add option to assign location(s) for additional resourcesAdd enhancement to support Locations for system services.

AA
Auth Codes
Call Park
Call Queue
Conferences
Hunt Groups
Paging
others...

Rename Location to Server in Call Park page
EnhancementLocations
SIPX-318Users in different locations are allowed to call each other by defaultUsers in different location should not be allowed to call each other by default.

1.Created two Locations.
2.Created two user groups with one user in each group.
3.Assigned a different location to each user group.
4.Verified in mongo that users are assigned to correct location.
5.Called between these two users
Result : users can successfully call each other.
Expected : they should not be able to
EnhancementLocations
SIPX-321Add more info on Locations pageBy default calling between different locations will not be allowed.
This could be important information to have in the admin gui either on the locations page or under the Add location page.
Perhaps there is more relevant info that should be added..
EnhancementLocations
UC-3763Change System -> Emergency Dialing -> Locations text box nameAs part of adding the new Locations feature we changed the 'Location' text box name to read "Emergency Response Location" in Emergency Dialing so as not to cause confusion with Locations.

Also modify description under text box that currently reads "Physical location details (such as room, floor, building, street address). This should be as fine grained as possible. Third party applications may use this to update the Public Safety database." to read "Physical emergency response location details (such as room, floor, building, street address). This should be as fine grained as possible. Third party applications may use this to update the Public Safety database."
EnhancementLocations
SIPX-142Make TCP default transport for Polycom phonesIn many cases recently it has become apparent that continuing to use UDP for SIP signalling with Polycom phones hasn't really worked:

* Server Failover in HA environments
* Large BLF lists

TCP is also inherently more reliable which results in a more consistent customer experience.

This ticket is for two configuration items:
* Make TCP the default transport protocol for Polycom Phones
* Enable TCP Fast Failover by default
EnhancementPolycom
SIPX-275Polycom Configuration improvementsAdded some new configuration features to the Polycom configuration plugin.

- Added ACD Agent login-logout
- Added admin and user password configuration
- Added labels for Polycom firmware 5.2.1, 5.2.2, 5.2.3, 5.3.0, 5.3.1, 5.4.0
- Added configuration for local processing of DND and FWD, when server-based feature control is active
- Making device backgrounds editable (SoundPoint and VVX series) with enable/disable feature

Tested with 15.10 and VVX 300/310/410/500
EnhancementPolycom
SIPX-329Configuration option for Push and State Polling User, Password and TypeAdded the configurable parameters for Polycom plugin so that user and password for push and polling messages could be configured.

Used for remote control and tracking of phone status

Tested with 15.10, Polycom VVX 500, VVX 410

Parameter on the phone
WebGUI of VVX Phone > Settings > Application > Phone State
WebGUI of VVX Phone > Settings > Application > Push

Parameter in config gui
Devices > Phone Group > Polycom VVX... > Applications

Moved UCDesktop in Menu Applications
EnhancementPolycom
SIPX-88Polycom firmware 5.1.1 default administrator passwordAdded the ability to set the Admin user and password for Polycom phones.

After upgrading to Polycom release 5.1.1 rev B a warning message is displayed to the user: "Default admin password is in use, please contact your administrator"
EnhancementPolycom
UC-3244zen 3420 : Desktop integration missing from VVX 1500 templateAdded the desktop integration portion of the Polycom template is for the VVX 1500 template, whereas it is there under all other VVX templates.EnhancementPolycom
UC-3756Update Polycom - ezuce custom background imagesUpdated eZuce background images that Polycoms load on their displays.

Update the ezuce logo to use the new one and change color scheme - currently background is white (this is an issue because the menu options are also written with white).

VVX400 does not have custom bg (works for vvx410)

VVX600 needs to have the ezuce logo differenty - currently on the main screen , you cannot see the ezuce logo because it is blocked by the black bar where softkeys reside.
EnhancementPolycom
SIPX-323Yealink ACD and SIP Server TypesAdded some new configuration parameters for Yealink Phones:

ACD inital state on login
ACD auto login after reboot

SIP Server types
UCAP, Genesys and Genesys Advanced
EnhancementYealink
UC-3714Internal Error when Adding PSTN to Audiocodes Mediant 1000 GatewayFixed an issue when attempting to add a PSTN Line to an Audiocodes Mediant 1000 PRI gateway profile the error below is presented in sipxconfig.log.

Caused by: org.springframework.orm.hibernate3.HibernateObjectRetrievalFailureException: No row with the given identifier exists: [org.sipfoundry.sipxconfig.gateway.FxoPort#-1]; nested exception is org.hibernate.ObjectNotFoundException: No row with the given identifier exists: [org.sipfoundry.sipxconfig.gateway.FxoPort#-1]
at org.springframework.orm.hibernate3.SessionFactoryUtils.convertHibernateAccessException(SessionFactoryUtils.java:677)
at org.springframework.orm.hibernate3.HibernateAccessor.convertHibernateAccessException(HibernateAccessor.java:412)
at org.springframework.orm.hibernate3.HibernateTemplate.doExecute(HibernateTemplate.java:411)
at org.springframework.orm.hibernate3.HibernateTemplate.executeWithNativeSession(HibernateTemplate.java:374)
at org.springframework.orm.hibernate3.HibernateTemplate.load(HibernateTemplate.java:551)
at org.springframework.orm.hibernate3.HibernateTemplate.load(HibernateTemplate.java:545)
at org.sipfoundry.sipxconfig.gateway.GatewayContextImpl.getPort(GatewayContextImpl.java:76)
at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:57)
at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:43)
at java.lang.reflect.Method.invoke(Method.java:606)
at org.springframework.aop.support.AopUtils.invokeJoinpointUsingReflection(AopUtils.java:317)
at org.springframework.aop.framework.ReflectiveMethodInvocation.invokeJoinpoint(ReflectiveMethodInvocation.java:190)
at org.springframework.aop.framework.ReflectiveMethodInvocation.proceed(ReflectiveMethodInvocation.java:157)
at org.springframework.orm.hibernate3.HibernateInterceptor.invoke(HibernateInterceptor.java:111)
at org.springframework.aop.framework.ReflectiveMethodInvocation.proceed(ReflectiveMethodInvocation.java:179)
at org.springframework.security.access.intercept.aopalliance.MethodSecurityInterceptor.invoke(MethodSecurityInterceptor.java:64)
at org.springframework.aop.framework.ReflectiveMethodInvocation.proceed(ReflectiveMethodInvocation.java:179)
at org.springframework.aop.framework.adapter.MethodBeforeAdviceInterceptor.invoke(MethodBeforeAdviceInterceptor.java:52)
at org.springframework.aop.framework.ReflectiveMethodInvocation.proceed(ReflectiveMethodInvocation.java:179)
at org.springframework.aop.framework.JdkDynamicAopProxy.invoke(JdkDynamicAopProxy.java:207)
at com.sun.proxy.$Proxy32.getPort(Unknown Source)
at org.sipfoundry.sipxconfig.site.gateway.port.PortSettings.pageBeginRender(PortSettings.java:65)
at org.apache.tapestry.AbstractPage.firePageBeginRender(AbstractPage.java:409)
at org.apache.tapestry.AbstractPage.renderPage(AbstractPage.java:244)
at org.apache.tapestry.engine.RequestCycle.renderPage(RequestCycle.java:400)
... 69 more
Caused by: org.hibernate.ObjectNotFoundException: No row with the given identifier exists: [org.sipfoundry.sipxconfig.gateway.FxoPort#-1]
at org.hibernate.impl.SessionFactoryImpl$2.handleEntityNotFound(SessionFactoryImpl.java:435)
at org.hibernate.event.def.DefaultLoadEventListener.load(DefaultLoadEventListener.java:233)
at org.hibernate.event.def.DefaultLoadEventListener.proxyOrLoad(DefaultLoadEventListener.java:269)
at org.hibernate.event.def.DefaultLoadEventListener.onLoad(DefaultLoadEventListener.java:152)
at org.hibernate.impl.SessionImpl.fireLoad(SessionImpl.java:1090)
at org.hibernate.impl.SessionImpl.load(SessionImpl.java:985)
at org.hibernate.impl.SessionImpl.load(SessionImpl.java:978)
at org.springframework.orm.hibernate3.HibernateTemplate$3.doInHibernate(HibernateTemplate.java:558)
at org.springframework.orm.hibernate3.HibernateTemplate.doExecute(HibernateTemplate.java:406)
... 92 more
FixAudiocodes
SIPX-293Configuration fails to deploy due to IVR exceptionFixed an issue with a java.lang.NullPointerException when backup server was set to a different IP.FixBackup
SIPX-331Call Queue service labeled as "beta"Remove the (beta) tag on Call queue in Admin Portal.

Go to System -> Services -> Telephony Services "Call Queue (beta)" service should be renamed to "Call Queue".
FixCallQueue
SIPX-274Conference maximum legs field , description updateFixed an issue that when creating a new conference, the "Maximum legs " field is set by default to "0",
and has the following description : "The maximum number of call legs to be allowed by this bridge. 0 means unlimited."

Reported problem : Setting up "1" in the "Maximum legs" field will have the same effect as setting "0" - unlimited users will be able to join the conference.

*Setting up "2", will permit two users to join the conference,then the 3rd one will be asked to enter PIN,and upon entering he will hear a prompt "Locking conference" and get disconnected.F
FixConference
UC-442Conference Control Labels incorrectFixed an issue on the conference config page where the DTMF digits for various conference operations, the "Volume" and "Microphone Gain" options are defined as they were labeled backwards. For example, a conference participant presses "6" to increase his volume into conference or "3" to increase the conference's audio to him. But the labels read the opposite.

The Freeswitch config the GUI generates the following:

<control action="vol talk up" digits="3"/>
<control action="vol talk zero" digits="2"/>
<control action="vol talk dn" digits="1"/>
<control action="vol listen up" digits="6"/>
<control action="vol listen zero" digits="5"/>
<control action="vol listen dn" digits="4"/>

The labels here seem to correspond to the GUI labels, "vol talk" for Mic Gain and "vol listen" for Volume. That makes sense. The actual behavior, however, is opposite.
FixConference
SIPX-114Translations in the login screen don't workFixed an issue with the login to the admin GUI so that it would use the default system language.

When the browser default language is for example Dutch and translations for the login screen are available they won't be used in some cases, the text will still be in English.
FixConfig
SIPX-240Creating Snom Group results in Error screenFixed an issue with phone groups other than Polycom.

It was reported initially that when adding a group for snom phones - snom320 in this case, but same with any of the others as well.

Select snom320 from list of devices. Make a change in the group - select save - go to error screen with the following - An internal error has occurred. Click here to continue.

When you go back to the group, your changes are saved in the profile.
FixConfig
SIPX-283sipXcom version fails to start when installing from RPMsFixed an issue where cfengine fails to run due to missing local_cp_by_digest bundleFixConfig
SIPX-328Sipxcom 15.08 Install Errors using Centos minimal MethodFixed an issue caused when user ran 'yum install sipxcom' instead of 'yum install sipxecs'.

Workaround for 15.08 - use 'yum install sipxecs'.

Installed Centos minimal and then installed Sipxcom using this methodhttp://wiki.sipxcom.org/display/sipXcom/Installing+on+Fedora+and+CentOS.

The following errors were generated following execution of sipxecs-setup. System is unstable and cannot be used.

Centos installation method is used for remote locations where transmission of ISO files is easier with Centos ISO than Sipxcom ISO because image is smaller.

Application settings:
Primary server : yes
Host : pbx
SIP Domain : sipxtest.com
Network Domain : sipxtest.com
Would you like to change your application settings? [ enter 'y' or 'n' ] : n
Initial setup, this may take a few minutes...
error reading information on service vsftpd: No such file or directory
error reading information on service vsftpd: No such file or directory
error reading information on service vsftpd: No such file or directory
error reading information on service mysqld: No such file or directory
error reading information on service mysqld: No such file or directory
error reading information on service mysqld: No such file or directory
Finishing system configuration, almost done...
error reading information on service vsftpd: No such file or directory
error reading information on service vsftpd: No such file or directory
error reading information on service vsftpd: No such file or directory
error reading information on service mysqld: No such file or directory
error reading information on service mysqld: No such file or directory
error reading information on service mysqld: No such file or directory
error reading information on service xinetd: No such file or directory
error reading information on service xinetd: No such file or directory
error reading information on service xinetd: No such file or directory
done.
FixConfig
SIPX-50When saving a group onSave() is triggered twiceFixed an issue that caused group settings saves to be done twice.

The problem was that when calling from UI we did:

Group group = getGroup();
if (equalsIgnoreCase(User.GROUP_RESOURCE_ID, group.getResource())) {
getCoreContext().saveGroup(group);
} else if (equalsIgnoreCase(Phone.GROUP_RESOURCE_ID, group.getResource())) {
getPhoneContext().saveGroup(group);
} else {
getSettingContext().saveGroup(group);
}

In CoreContextImpl we have:

@Override
public void saveGroup(Group group) {
m_settingDao.saveGroup(group);
}

both methods were intercepted and onSave event was published.
FixConfig
UC-2407Error message after changing User ID needs an updateFixed an error message to indicate that Password needed to be changed and not PIN.

Steps to reproduce:
1.Go to Users->Users->User 200 and changes its User ID to 250
2.Apply

Reported issued : you will receive an error :
"When changing user name, you must also change PIN"
Expected : When changing user ID , you must also change Password
FixConfig
UC-2763Remove IM Chat Room section from Conference pageWith UC-2671 Decouple Conferences and Chat Rooms - getting fixed, the IM Chat Room section is no longer needed under the Features->Conferencing->"Server"->Conferences->"Conference"

These settings were removed.
FixConfig
UC-3228Linksys spa942 phones need template changeFixed an issue where for mwi the voicemail box (mailbox id) should be set to the user extension and voicemail server should be blank. If many of these phones were configured the old setting caused a flood of subscribes to the server.FixConfig
UC-3596Add nodes IP and FQDN in /etc/hostsWe received some tickets where although named is starting prior to mongod it fails to serve IP address of the other nodes resulting in following errors on mongod.log. We've added server node names and IP addresses to servers' /etc/hosts file to alleviate the problem.FixConfig
UC-3663eMail notification checkbox has opposite effectIt was reported that when you did not check the box to notify the user of account creation that the user was notified.

When you add a new user , the NOTIFIED (and not "notify" - there's some confusion) checkbox will be by default unchecked.
That means the user will be informed by his user creation
Otherwise if you check it , the system considers the user to have already been notified , so the email is not sent.

We changed the label from "Notified" to "User was notified".
FixConfig
UC-473Change LDAP + PIN to LDAP + PasswordFixed an issue with error on login where the error message indicated PIN, it should reference Password.FixConfig
UC-492Phone Book not Paginating CorrectlyFixed a problem with the ‘Next’ and ‘Previous’ page buttons not working (remain greyed out) regardless of the count of users within the phone book as best as possible with the control application that was used for this feature. If we limited pagination to 12 the search feature would only search those 12 for a user. We decided to set the page size to 500 to keep searching working. Next and Prev will work if you have over 500 users in phonebook.FixConfig
SIPX-313References in from settings_with_location table not removed when node is deletedFixed an issue that caused the settings_with_location table is not cleaned up when locations deletedFixLocations
SIPX-278MoH not working when phone behind NATFixed an issue with remote phones would not get MoH.

This does not replicate on setups where servers and phones are on the same network.(no NAT)

Configuration:
Server is installed in AWS (behind NAT)
Phones are behind their own NAT.

Added two set of logs.
In one set, call is between two Polycom phones (fw 4.1.7) - there is no MoH.
In the other set,call is between a Polycom and a Bria softphone - there is MoH only when Polycom holds the call.

Fail scenario <Polycom phones>--Nat--Internet--Nat--<openUC> :
1.Polycom 10.2.0.51 - extn 2801 calls Polycom 10.2.0.52 - extn 2802
There is voice path both ways
2.2801 holds call
Result Fail : 2802 does not hear Moh
(also tested when 2802 was the one to hold the call,there was no moh as well,but i did not include that in the logs/capture)

Semi-Fail scenario <Polycom + Bria >--Nat--Internet--Nat--<openUC> :
1.Polycom 10.2.0.51 - extn 2801 calls Bria 10.2.0.200 - extn 2804
There is voice path both ways
2.2801 holds call
Result Success : 2804 hears Moh
3.2801 unholds call
There is voice path both ways
4.2804 holds call
Result Fail : 2801 does not hear MoH
FixMoH, NAT
SIPX-243Polycom Configuration Server Feature Control DND/FWDFixed an issue where the Polycom Plugin generates wrong Parameter in Server Feature Control for DND and Forward

Parameter is
serverFeatureControl.doNotDisturb.enable
not
serverFeatureControl.doNotDisturb.enable.enabled

Same for Forwarding

Related to SIPX-36
FixPolycom
SIPX-36DND disable does not workFixed an issue that caused the Polycom disable DND feature to not work.

There was a typo in the syntax that is used in the config file generated for each phone.
The config file contains:
feature.doNotDisturb.enable.enabled="0"
But it needs to be
feature.doNotDisturb.enable="0"

This is controlled by the setting under Features > Do Not Disturb in the phone or group config.
FixPolycom
SIPX-78Polycom VVX 1500 Missing Desktop IntegrationThe Polycom VVX 1500 phone did not have the Desktop Integration settings available in the template. Added the settings.FixPolycom
UC-2305GUI crash in a regions partitioned setup (intermitent)Fixed an issue with Regions such that when Cluster was in a partitioned state after making config changes that apply to all servers, GUI would crash because httpd service crashed.

Setup : 8 servers in HA with 3 regions (2+1 default)

Steps :
1.Default region is no longer available - partitioned
2.From the GUI make a config changes that applies to all servers

Reported issue : After making config changes that apply to all servers, GUI will crash, httpd service will crash

Workaround : service sipxconfig restart
FixRegions
UC-3750GRUU problem in 15.06Fixed an issue with GRUU caused by an earlier commit (introduced in 15.06) that caused GRUU to break when used with wss transport.

Customer was using GRUU address to target specific registration instances.
Customer used to receive the contact's GRUU address as:
~~gr~BtXVZtl2D-l@server.fqdndomain.com;gr
and now doesn't receive it on registration.
FixSIP
SIPX-310Problem receiving faxes in 15.06 and later & sipXbridge Specs ViolationFixed an issue with sipXbridge that caused certain SDP values to not be passed. This was manifested by a FS but that's adding additional SDP settings.

During a T.38 re-INVITE sipXbridge strips a custom payload from the m line sent by FS. The resulting m-line has no payload at all which of course is a syntax error. This seems to be a direct result of the regression pointed out here:https://freeswitch.org/jira/browse/FS-6954.

This may cause SBC's that don't pass the m-line in SDP to not be able to receive faxes.
FixSIPTrunk
UW-313Notifications no longer workingFixed an issue that caused the red dot representing new IMs to no longer be displayed on the browser tab.
Sound notifications also don't work.
This is the Console error:

TypeError: Notify.isSupported is not a function
at Notify.myNotification.needsPermission (fc9f128370b1ea9770ccfadb.scripts.js:1)
at fc9f128370b1ea9770ccfadb.scripts.js:1
at handleMessage (fc9f128370b1ea9770ccfadb.scripts.js:1)
at Object.h.Handler.run (4e42ce69bdd61567355bfd36.modules.js:3)
at 4e42ce69bdd61567355bfd36.modules.js:3
at Object.h.forEachChild (4e42ce69bdd61567355bfd36.modules.js:3)
at Object.h.Connection._dataRecv (4e42ce69bdd61567355bfd36.modules.js:3)
at Object.a.Bosh._onRequestStateChange (4e42ce69bdd61567355bfd36.modules.js:4)
FixUniteWeb
SIPX-307Yealink configuration crashes on applyFixed an issue with Version 15.06 - 15.10 that caused the Yealink Configuration to crash on apply (similar to SIPX-240)

Failure occurs in "org.sipfoundry.sipxconfig.site.phone.EditPhoneDefaults.java), inside the apply Method the "DeviceVersion" throws a null-pointer exception.

The DeviceVersion is null, because inside the Yealink-Plugin the a phone is tagged with "yealinkPhone" and the configuration looks for "yealink"

To handle this, it is necessary to change "yealinkPhone" to "yealink" to make it work (plugin folder and bean_id)

Made some changes to do this (pull request follows), but this means the bean_id will changes, so older entrys in the database will crash the web GUI.

To fix this a change inside the postgres is needed, which is not part of the commit source. Tested with 15.10.

SQL Statement to fix after update
UPDATE "public"."phone" SET "bean_id"='yealink' WHERE ("bean_id"='yealinkPhone')
FixYealink
SIPX-309Yealink DND mode.Fixed a problem with DND mode on Yealink phones.
"Feature Key Synchronization" - must be unchecked in phone's web interface or by autoprovision fille.
FixYealink
SIPX-332Polycom VVX 5.4.0 only supporting 34 total linesIt was reported that Polycom firmware 5.4 supports 34 total lines+BLFs compared to 48 on firmware 4.1.7

e.g. On VVX 600 running fw 5.4.0 if you deduct the total number of line keys supported, which is 16 , you will only have 18 slots available for BLFs
Compared to 4.1.7 - 32.

Also valid for firmwares 5.3,5.2,5.1


From: http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/EM_AG_Addendum.pdf

Polycom phones can have multiple registrations; each registration requires an address or phone number. All VVX phones support up to 48 registration line keys and up to 34 line registrations when connected to three expansion modules.
Known IssuePolycom