freePBX 2.2 Release Candidate 3

IF you are still on beta 1, 2, 3, RC1 or RC2 PLEASE UPGRADE

This is to find and fix as many bugs as possible. There are, however, still some issues that may or may not be fixed before 2.2 is actually released. If you find a bug, or something that's not working as the documentation says it should, please first check below to see if it's already reported, and if not, create a new ticket.

A more detailed 2.2beta and release candidate Bug Report can be seen on this link as well.

All tickets that you reported (not just 2.2) can be seen on this link. (You must be logged in though.)

Known Issues

Full list

Closed Issues

#891

Implement extension/device specific dial options

#1031

inventorydb and customerdb not working under 'tools'

#1055

Limit the number of incorrect password trys for a meetme confrence

#1056

Unable to delete an inbound route

#1067

custom extenstions bug creatings speed dials

#1094

gui element section headings not added in order created

#1119

Transferring a queue-originated call to another extension bypasses voicemail

#1122

New recordings are not saving after upgrade

#1148

Music on hold options with Queues

#1156

*extension transfer to voicemail does not tell who the person is

#1161

PHP errors after updateing to SVN 2645

#1179

engine_getinfo() does not work in web interface

#1182

can't use # transfer when call is outside number

#1185

he_IL translations for all the rest of the modules not translated in #1153

#1189

System Recordings

#1196

Transfers from a queue agent do not ring through to voicemail

#1205

#999 Return of the Undef

#1208

freepbx displays the manager password on the gui

#1222

Error upgrading module asterisk-cli

#1223

Module Admin menu item no more in Tools section

#1224

Category based menu and extensions vs device/users

#1225

Category based menu and permissions

#1226

index.php - Fatal error

#1239

Module QuietMode option does not present a blank background

#1240

Asterisk-CLI HTML error

#1241

Logfiles module HTML error

#1242

Jave SSH Module HTML bug

#1243

IVR HTML bug

#1244

System Recording Module HTML Bug

#1254

Error in extensions changing privacy manager setting

#1255

Misc Destinations list does not conform to new look

#1256

Ring Groups list does not use new list look

#1257

Time Conditions List does not use new format

#1258

Follow Me list does not use current html list format

#1259

Conference list does not use current html list format

#1260

IVR list does not use new html list format

#1261

On Hold Music list does not use new html list format

#1262

Pin Sets does not use new html list format

#1263

Paging Module does not use new list format

#1264

Queues Module does not use new html list format

#1265

System Recordings does not use current HTML list format

#1266

Callback module does not use new html list format

#1267

DISA module does not use new html list format

#1268

Caller Name Lookup module does not use new HTML list format

#1269

If all agents are busy, caller can access agent login

#1279

no defaults in ampbackup.pl and schedule_function.php error message with weekday

#1280

Gabcast Module not upgrading

#1281

Dictation not working

#1285

Use of HTTP_SESSION_VARS in CDR scripts breaks in PHP 5

#1286

PHP 5 uses a different get_headers

#1288

VMWI

#1289

*80 *54 *55

#1290

Dication does not send email.

#1291

Gabcast module doesn't allow adding a channel.

#1292

1275 Call to undefined function core_users2astdb()

#1294

user CID not overriding trunk CID (outgoing)

#1296

php error after upgrade from amp 1.10.010

#1297

meeting room configuration not available after upgrade from amp to freepbx

#1298

error while running upgrade script

#1299

missing lines in SQL/newinstall.sql file

#1300

error creating voicemail options when creating new sip extension

#1302

Confirm Calls no longer works in Ring Groups

#1304

context=from-truck is added to the wrong place in sip_additional.conf

#1305

Uploading MOH files makes very long file name

#1307

music on hold not enabled after upgrade

#1308

Unnamed IVR created

#1309

Problem with reports

#1310

CID Source not selectable in inbound routes

#1311

Queues don't work if one static agent is on DND

#1312

Uploading MOH

#1313

After User Deletion, Incomplete Submit Page Displayed

#1314

Apply configuration changes does not work, and is always there

#1315

Follow Me information not deleted when User Delete

#1319

Zap Barge 888 hearing beeptone

#1320

extension outbound cid

#1322

Upgrade to Beta2 from 2.1.3 - CDR Analyser

#1323

Putting format <1234> within display name breaks internal callerid

#1324

sip_additional.conf not saved when add trunk or extension

#1326

Bug in voicemail options in Extension setup

#1327

New UI broken under FF 1.0.7

#1328

CustomerDB module gives error when clicking on it.

#1330

HudLite not Dialing X-Lite Softphone

#1331

java script capability not being recongnized

#1332

cid lookup bug

#1334

Dictation codes *34 and *35 only get invalid extension recording

#1335

SQL error when adding administrators

#1336

Cannot ring queue

#1337

app-dictate uses bogus message "invalid"

#1338

app-dictate uses bogus message "invalid"

#1339

when recording via extension the beep should not be recorded

#1342

Voicemail Options Are being corrupted in the display

#1345

freepbx.png is too large for all of the web portal pages

#1346

Red bar not updating on IE6

#1349

No voicemail radial button defaults and/or allows a submit without some set causing corruption

#1351

Unable to dial ring group or queue from IVR

#1352

when selecting confirm call my phone rings, but no message is played and it then hangs up

#1353

Don't assume AMPCGIBIN

#1357

Add text to Ring Groups "Extension List" popup

#1358

In Follow-Me, if "Remote Announce" is set to None, remote calls get disconnected

#1359

Hunt behaves like ring all in ring groups

#1360

Module Hooks Are Not working right (is it cause of gui elements?)

#1361

Mailbox not being setup properly

#1362

Extension (and probalby User) display is completely out of order

#1363

paging module completely broke

#1365

Changing "mailbox" setting in extension context has no effect (trying to share a common mailbox for two extensions)

#1371

Default music-on-hold files always get installed

#1372

Play back caller ID in macro-confirm

#1376

Can not create or modify Extensions, using Core 1.2

#1378

Voicemail not working with Ring groups or Followme

#1379

CID Lookup Bug (still there)

#1381

Paging And Intercom 1.5 Broken?

#1383

Problem restoring from freepbx backup module

#1384

Recording select boxes are indexed by number, not filename

#1387

User's voicemail "unavailable" recording is not played in many circumstances

#1390

Fatal error with Parking Lot

#1391

Setting CID Lookup Source in Incoming Route doesn't "stick"

#1392

Setting CID Lookup Source in Incoming Route doesn't "stick"

#1394

CID sent to Wakeup Call is Outgoing Caller ID

#1395

Documentation or Instruction Out of Date.

#1396

Hints are hard to read on IE

#1397

Module update leaving multiple httpd processes running

#1398

PIN not saved in Trunk definition

#1399

Random Play Breaks MOH

#1400

IE6+ and PopUps

#1402

Caller ID Lookup and Inbound Routes

#1405

Hanging Server!! using features.conf settings of ## or basically anthing other then # cause server to go into hand

#1406

voicemail creation problem

#1408

Pointng calls to "Core: voicemail box" destination - VM greeting not being played

#1410

beta3 and IAX extension...

#1411

DND not getting the IAX extension number

#1413

Inbound route definition for Zap channels intermittant

#1414

Speak your exten number doesn't speak the extension

#1415

Right Nav bar still has Next Previous (needs removing if scroll bar)

#1416

internal callerid not being set to "Display Name" and defaulting to device on new extensions

#1417

IVR doesn't use reload() function

#1419

Announcement module should depend on 2.2beta3

#1422

Backup not working

#1423

Followme/RG confirm call fails if Asterisk Dial command options are blank

#1424

phpagi.php get_variable strips off trailing parenthesis in arguments

#1427

Voicemail option not showing in extension setup window

#1429

Report does not show Duration correct on second page

#1430

RTP packets not using NAT IP address, but SIP packets are

#1431

Call to custom ext formated SIP/Num@domain doesn't work in beta 3

#1432

PINs Brocken

#1433

If two extensions have call forward on busy pointed at each other, incoming call causes runaway condition.

#1435

privacy.conf minlength=xx ignored

#1439

Extensions numbered in 300's can't receive incoming calls

#1444

unable to save the callerid lookup sorce on the inbound route

#1448

Lost Pinset

#1449

Don't page phones that are in use

#1452

"Backup & Restore" module upgrade fails

#1453

Extensions Class Typo Bug

#1454

Reload after restore restores again instead of reloading

#1456

Paging module broken after 2.2.0RC1 upgrade

#1457

SellVOIP auto-config blows up

#1458

Incorrect message text adding an extension

#1459

No ToolTip text for AddRingGroup->GroupDescription

#1461

RC1 - # does not get directory, instead it disconnects.

#1462

RingGroup will not allow you to select any other ring strategy then ringall

#1463

Modules on 2.2.x (including b3, rc1) Upgrade All doesn't work

#1464

Error when deleting a MeetMe Conf

#1465

Unable to call a newly created MeetMe Conf

#1466

Upgrade Fails from 2.1.1

#1467

UI problem in Firefox 2.0 after upgrade to 2.2.0RC1

#1468

Missing voicemail/dictation options when creating Users

#1469

Default user not created when adding devices

#1470

SellVoip module

#1472

default manager password warning

#1473

Invalid Predefined Patterns in Outbound Routes

#1475

Cannot register Cisco 7940 running SIP 7.5

#1476

MOH Upload Problem

#1477

Queue calls recording/monitoring not working, ARI and Reports don't show them

#1478

Module Admin > Upgrade, does so out of sequence

#1480

Adding Extension without VM-Password causes "ann error occured" instad of "busy"-signal

#1481

"Direct DID" in extension configuration doesn't do anything

#1485

javascript Validation Broken with guielements (so it seems)

#1488

No ampbackup.pl installed

#1489

"Error occured" when voicemail disabled

#1490

SetCallerPres(prohib_passed_screen) is not working (Patch)

#1491

DID and not CID is used with Phonebook dial-by-name directory

#1493

PHP5 Variable Definitions

#1499

Never Override CallerID Works...Backwards

#1500

Never Override CallerID Works...Backwards

#1501

Never Override CallerID Works...Backwards

#1502

Calls through IVR to an extension defined in extensions_custom.conf no longer hear ringing in RC1

#1503

CID incorrectly set on new/modified extensions

#1504

queue not distributing call properly

#1505

incoming fax not working when using "from-zaptel" context for incoming zap calls

#1507

Queue Agent Announcement

#1508

Dynamic agent not working. Phones not ringing, but FOP show yellow LED.

#1509

MOH - Random Play does not randomly play music files.

#1510

Press 1 to listen to your recording doesn't work when creating a System Recording

#1511

Flash Control Pannel and Zap channel display

#1512

Backup and restore 2.0.1 does not work

#1513

Inbound Recording stops when transferred

#1514

Incoming CID does not pass when transferred

#1515

Congestion() should be called with an argument, ie Congestion(20)

#1516

dgettext() calls break when gettext extension not loaded

#1518

online module check for updates gives error

#1524

HTTP Caller ID lookup doesn't work

#1525

Congestion and Busy should have a argument (again...) (extensions.class.php)

#1526

Paging broken, hangup after 5 seconds

#1527

amportal start and safe_opserver start FOP in an infinate loop

#1528

setup for amd 64

#1531

Busy Line Field Does Not Work on Aastra phones

#1534

I am having the same MOH problem as Ticket #1476

#1535

IVR

#1537

MOH not propogating table on new install

#1540

Sip extension port or bindport

#1541

RC1 and *1.4B4 does not create "sip_additional.conf" or "iax2_addinional.conf"

#1543

Use of AMPADMINLOGO

#1545

Use _X. warning

#1547

Custom contexts/extensions not working

#1549

Mistake in /admin/functions.inc.php drawselects()

#1551

Memory Hunt not working or intermitent on Ring Groups

#1552

Voicemail cannot be enabled for User

#1555

NO IVR Options (same as ticket 1535)

#1557

Queue handling for serious call centres

#1559

When IP address changes the panel will not work.

#1563

Error when trying to Check for Update Online

#1566

using # key for transfer hardware crash the PC

#1570

Number of seconds to ring phones before sending callers to voicemail doesn't work

#1575

Incoming FAX don't work on freePBX 2.2rc1

#1576

Join Announcement in Queue keeps on repeating

#1577

CID not displayed in Queue

#1578

Destination on inbound routes not saved

#1580

Join Announcement not working with queue

#1581

Misc Applications module doesn't trigger red activate ba for custom-apps

#1584

Weirdness in Parking Lot behavior (10 second delay, MOH disappears, ZOMBIE lines in CLI)

#1589

delete in misc applications module produces an error

#1593

Upgrade All will sometimes install new modules instead of just upgrades

#1594

Main screen still shows rc1 instead of rc3

#1595

Saving of duplicate feature codes not allowed (cannot be overridden for use in custom contexts)

#1596

Create System-wide speed dials?

#1599

IVR

#1601

amportal stop never kills the panel

#1604

Ring Groups

#1605

Apply changes not writing configs

#1606

(Destination) in (Inbound Routes) is not displayed when created or edited

#1607

dbGet and dbPut has been removed in asterisk 1.4

#1609

Conference - Music on Hold.

#1616

MOH problem deleting a class

#1617

Music on Hold disappears and CLI contains <ZOMBIE> when transferring from Linksys/Sipura device

#1619

Online upgrade on the Misc Applications 0.1 module doesn't work

#1621

Extension Caller-ID overrides trunk Caller-ID setting

#1624

Setting CID in both the trunk and extension results in an Unavailable CID to recipient

#1625

softfax patches broken

#1627

Asterisk conf files not updated

#1629

freepbx shouldn't silently change bxfer key

#1630

Asterisk Directory file name incorrect

#1631

Music on Hold Randomization

#1634

Upgrade from 1.x to 2.2.0rc3 breaks dial plan by forcing fax and insecurity

#1636

misspelling of 'version' as 'verison'.

#1639

Dial rules not backed up with Backup/Restore Module

Downloading 2.2

• Sourceforge Download page

There are two releases - freepbx-2.2rc3.tar.gz, and freepbx-2.2rc3-withmodules.tar.gz. The 'withmodules' one contains all the freePBX modules. This is for ease of installation on machines that don't have full internet access, and for new installations, to save you downloading all the modules individually. In most cases, people upgrading do not want to get the 'withmodules' one, whilst a new install almost always does want to get the withmodules.

Note that modules are often updated online, so even if you have got the 'withmodules' package, there may be module updates availble online that were released after the 'withmodules' package was created.

Getting The Latest 2.2 beta and RC Fixes

The online module upgrade system will pull the latest modules when they have been fixed. However, there are frequent changes to the core system during this beta that will not get updated and will not be put into the tarball described above until another one is created (which will probably be ever 2-3 weeks).

You can keep up-to-date and get fixes to closed bugs by pulling and keeping up to date with the svn head of the 2.2 branch, which is limitted to bug fixes only and will become the main 2.2 branch when we release.

To export the most recent 2.2 beta branch, do the following on your system and use the generated directory to update your system:

• svn co https://svn.sourceforge.net/svnroot/amportal/freepbx/branches/2.2 freepbx-2.2beta-latest

Once you have done that, you can simply do the following from within the freepbx-2.2beta-latest directory to update:

• svn update