Changeset 4014

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Timestamp:
06/08/07 20:52:54 (5 years ago)
Author:
p_lindheimer
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Merged revisions 3726-3727,3729-3732,3734-3735,3737-3738,3740-3741,3743,3745,3747,3752-3753,3756,3758-3769,3771,3773-3774,3778-3793,3797,3799-3804,3807,3809-3825,3827-3829,3831-3836,3838-3845,3847,3850,3852-3856,3859-3869,3871,3873-3882,3884-3885,3887-3890,3892-3895,3897-3902,3904,3906,3908,3910-3916,3918,3920,3922-3927,3929-3932,3934,3936-3937,3939-3946,3948-3955,3957,3959,3964,3967,3969-3970,3972,3974-3977,3979-3980,3982-3994,3996-4001,4003-4005,4007-4013 via svnmerge from
https://amportal.svn.sourceforge.net/svnroot/amportal/freepbx/branches/2.2

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r4011 | p_lindheimer | 2007-06-08 09:59:18 -0700 (Fri, 08 Jun 2007) | 1 line


update 2.2.2 CHANGE log with highlight of changes

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r4012 | p_lindheimer | 2007-06-08 12:40:20 -0700 (Fri, 08 Jun 2007) | 1 line


Creating release 2.2.2

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  • freepbx/trunk

    • Property svnmerge-integrated changed from /freepbx/branches/2.2:1-3724,3726-3727,3729-3732,3734-3735,3737-3738,3740-3741,3743-3745,3747,3752-3753,3755-3793,3797-4002 to /freepbx/branches/2.2:1-3724,3726-3727,3729-3732,3734-3735,3737-3738,3740-3741,3743-3745,3747,3752-3753,3755-3793,3797-4013
  • freepbx/trunk/CHANGES

    r3272 r4014  
    1 2.2 Final 
     12.2.2 
     2- To Get Full Details - look at the SVN logs of changes since the previous 
     3  release. These are only higlights. 
     4- WARNING: 
     5  merge ext-did and ext-did-direct all into ext-did context, and create 
     6  new context, ext-did-cacthall when creating an ANY/ANY route. The inclusion 
     7  of ext-did-catchall is in the extensions.conf file so if any customizations 
     8  have been done, make sure this is included. 
     9  The purpose of this change allows directdids specified with the extension 
     10  to properly co-exist with those create with inbound routing. In addition, 
     11  error checking has been added to keep the same did from being used two places. 
     12  However, you can use a did on an extension as a directdid, and then included 
     13  the same did+CID info on inbound routing and that is legal, and will now work 
     14  properly instead of being ignored as was the case in the past. 
     15- WARNING: 
     16  sip.conf file changes that MUST be adopted. The inclusion of sip_registrations.conf 
     17  and sip_registrations_custom.conf have been added to sip.conf. In the past the 
     18  registrations were put at the very top of sip_additional.conf which made it really 
     19  easy to break things if you put a custom sip context into sip_custom.conf. 
     20- javascript warning when users try to use the 'r' option in the 
     21  "Asterisk Outbound Dial command options" of the "General" tab. 
     22- allow the '=' character on the right side of an assignment in the trunk specification 
     23  section. This was a common error propblem if a secret included an '=' sign, for 
     24  instance. There are other settings that require '=' there also. 
     25- fix bug in ringgroups and followme when DND was enabled on the first extension of a 
     26  ringgoup, the others would not be tried. This behavior is correct if the ring 
     27  strategy includes the '-prim' postfix but was doing it to all strategies. 
     28- Added Israel and India Indications to General tab 
     29- Added MailBoxExists and WaitExten asterisk commands to extensions.class.php to enable 
     30  some bug fixes requiring those commands forthcoming to the IVR and Voicemail modules. 
     31 
     322.2.1 
     33- Fix ENUM lookup bug in 2.2.0 - r3546 
     34- Convert MP3 MOH files to SLN (Asterisk Native) format - r3548 
     35- module_install() now returns true for already installed modules - r3569 
     36- Allow null and blank values to be put into astdb - r3576 
     37- don't propogate dnd behavior and not ring other phones if this was not 
     38  a prim mode strategy - r3580 
     39- Apply fix for #1361 (patch #1667), mailbox field not being propogated in in 
     40  deviceanduser mode. - r3584 
     41- Fix typo in extensions.conf, when pushing '0' for oper and not having an 
     42  opereration extension defined, would pass a bad Dial string. - r3585 
     43- added warning on save of trunk if user context left blank and user details 
     44  filled in that details will not be saved #1666 - r3631 
     45- limit rnav width #1647  
     46  fixed panel displaying extensions over 9999 as trunks - ticket #1710  
     47  List device technology on page when editing Ticket #1711  
     48  fixed trunks stripping AMP: which removed ANY occurance of the letters  
     49  A,M,P,: from the beginning of all trunks, also unified the display on  
     50  the routing page - partially noted in #1713  
     51  CFB when dialparties.agi decides not to - offhook, user hits ignore/reject, 
     52  etc. - patch #1681 - (Backport from trunk) r3643 naftali5 
     53- now module_admin works even for "broken" modules, running from every  
     54  directory  - r3678 
     55- do not display warnings about password when not using mysql/pgsql - r3679  
     56- make the cdr page links a bit nicer - r3689 
     57- fix typo in sip.conf - r3691 
     58- keep rtone from being set in queues_additional.conf #1635 - r3697 
     59- fix queues retrieve conf bug part of #1659 - r3744 
     60 
     612.2 
     62- IMPORTANT CHANGE - Asterisk 'transfer' featurecode is now defaulted to '##', rather than '#'.  
     63  This was changed to avoid issues with sending a '#' to an externally called party. Note 
     64  that this is a SIGNIFICANT CHANGE, and you should be aware of it. 
    265- Fixed trunk 'locking' (Never Override CallerID) checkbox to work properly. This allows a 
    366  trunk to restrict outbound CallerID settings to those of the trunk or defined in an