PBX V3.68c
Before you start to run PCBest Networks SIP PBX v3, 
please you open sippbxv3.xml and configure your database connection first.


3.68c
1. Added SBC support for routing the call on different IP address


3.68b
1. Added agent dynamically login to ACD
2. Added PBX v3 Watch dog

Edit the file PBXv3Watcher.ini, 

Set pbxv3.caller.id and pbxv3.called.id(The most import is called id):

pbxv3.timer.value = 10000
pbxv3.caller.id = <sip:12443555@192.168.1.101>
pbxv3.called.id = <sip:8000@192.168.1.200>
pbxv3.max.num.not.connected = 1

pbxv3.timer.value is in milliseconds.  
pbxv3.max.num.not.connected is how many times call failed consistently.

Then create an inbound dialplan in PBX, for the call just goes to Music Server. 
Then set this number into called id, with the PBX's server IP address.

Start the service  PBXv3Watcher

Then on PBX, you will see the calls to it every 10 seconds. 


3.67 2020-04-23
1. Renamed column 'TaskID' of table auto_dialer_jobs and auto_dialer_done to 'JobID'.
A job id you can set and later retrieve the call record in auto_dialer_done table by using this id. It can be null.
2. Added column 'CallID' in auto_dialer_done table.
It is the same call id in CDR table to mark the call.

3.66c 2020-04-08
1. Included latest SDK for couple of bug fixes.
2. Fixed an issue with DuplexConnect

3.66 2018-02-21
1. Added security feature
2. Default disable RTP relay


3.65b
1. Integrated SDK with a recent patch for possible crash.


3.65a
1. Fixed a GUI bug with call black list. 
The bug was, when use added some rules in call black list and save, then the entries are gone after restarting the PBX.

2. Fixed a GUI bug with call back
The bug was, it errors when editing an existing call back rule.



3.64 & 3.65
1. Pin / Account Code
Added Pin code/Account Code for extension settings so user has to dial a certain PIN code to be able to dial out.

2. Paging and Intercom
Support paging for broadcasting messages

Currently it supports Snom phones, by using the following header:
Alert-Info: <http://www.notused.com>;info=alert-autoanswer;delay=0

or Phones which support Call-Info header for auto-answer:
Call-Info: sip:192.168.20.1\; answer-after=0

Other phones may need auto-answer is turned on.

3. Personal Voicemail box greeting recording
When accessing VM, added an option to record a personalized greeting via the handset.

4. Call Back
Added call back rule for caller numbers which don't answer and callback immediately. Very flexiable can customize 

5. Secondary SIP account for Outbound Dial plan in case the first SIP account is failing


3.63  2016-01-29
1. Added TCP support for SIP protocol.

3.62a
1. Syncronized SDK dll to fix an issue with call transfer

3.62
1. Fixed an issue with hold and unhold when client phone provide private IP in RTP address.
2. Fixed an issue with Monitor group number in Dialplan. PBX now checks monitor group number first.
3. Fixed an issue with extension 2 extension call log in exten_cdr table since 3.62
4. Added a field in agent table for agents to pause answering calls. Third party apps can use this filed to set.

3.61b
1. Added an option in system menu for stopping playing audio in IVR menu when first DTMF is pressed.
### As plugin interface also changed too in this release, so plugin must be rebuilt.###

2. Enhanced timeout(3s) for SQL queries.

3. Fix for delay answering when db response is slow. Created a new thread for taking care of db writing.

3.61a
1. Added GC colllect every 10 minutes for memory collection
2. Fixed a bug with writing cdr_exten acd data
3. Updated GTAPI.dll andGTSIPAPI.dll for a possible mixing call issue

3.61
1. Changed the column length of DiscReason for cdr_pbx, cdr_acd, and cdr_exten tables.

3.60a
1. Fixed a bug introduced by SDK for record-route not sending package to the route address
2. Resolved an issue with managerClient adding a channel multiple times to a conference room
3. Fixed a problem with call transfer drop not changing extension's status.

3.60
1. Added a feature for ring group, not answering call first before ring group destionation is connected
In ring group setting, now there is one more option to set if answer the incoming call first, then transfer to destion.
You can choose not for saving the cost of telephone line before really reaching destions.


3.59b
1. Added conference room status event for managerclient
public override void OnConferenceRoomStatus(string roomName, string channels)

2. Added a method to get the conference room status for manager client:
mc.GetConferenceRoomStatus(conf_name);

3. Solved an problem with rtpduplex on different audio codec.

3.59a
1. Added an option in server sip setting to set if Forward P-Asserted-Identity for call transfering.

3.59
1. Added an option to allow extension to append its extension id to specific caller did for sip account.


3.58
1. Added accept multiple call option for extension

3.57e
1. Added proxy only feature for extension to extension call
2. Added Message Text proxy for PBX

3.57d
1. Fixed a bug with blacklist which only takes the first one setting

3.57c
1. Adopted latest verison of SIP SDK
a. using memory pipe for events
b. fixed a private and public IP mix, and using public but no sound issue.


3.57b
1. Fixed a problem with conference crash.


3.57a
1. Integrated the latest SIP SDK for the following issues:
a. Enhanced with DNS SRV.
b. Fixed an issue with Send_MakeEx not setting DestIP and DestPort

Just try to enable:
CFG_SetValue("gtsrv.sip.enable.dns.srv", "1");

c. Fixed an issue with Nortel SIP phone device.
It has an extra tag in FROM:
From: <sip:6083@10.50.20.70:5060>;tag=a19c3028-11fa280a-13c4-55013-66aad4-28a124ff-66aad4;user=phone

Fixed similar problem for TO also.

d. fixed a problem with cancelling calls.

3.57
1. Added two new columns for auto_dialer_done table: FinalCode and FinalDesc, to give final response from remote for calls.
2. Integrated latest version of SDK DLL for fixing a problem of call cancelling.
3. Fixed an issue with moh.
4. Better support for Zoiper phones for holding call


3.56
1. Added support for mp3 format and gsm format for recording wav.


3.55
1. Added support to send extension, agent and channel status to managerclient side when connected
2. Added call time limit to extension, agent, dialplan and sip account.

3.54
1. Added an option in conference setting, to allow recording.
recording files for conference are saved into PBX\record\conference room\date\......

2. Updated cdr_pbx table, to add one more field for conference room.(ConfName)

3.53c
1. Added batch work for auto dialer task by support csv and txt file

CSV or TXT file format:
Use , or ; or | to seperate column.
TaskType,Caller,Callee,Date,UniqueKey(less than 12 characters)

For example(you can look at sample file AutoDialerSample.csv in PBX folder):
1,,16132652286,,BFY232364

Means:
Task type 1, caller is none, call to 16132652286, date is null(now), 
and unique key to mark this call job is BFY232364. Date format is yyyy-MM-dd HH:mm:ss.
You can search auto_dialer_done table for your unique key to find out call result for each job or call.

3.53b
1. Fixed a multiple local IP address and multiple SIP application running on different IP issue.

3.53a
1. Added one option in server/proxy site, to enable/disable SIP OPTION ping for proxy extension.

3.53
1. Added one column(CallID) in voice_mailbox to save the call unique id.
2. Added DiscReason column for cdr_ tables, for recording why the call is disconnected.
3. Used latest PCBest SIP SDK 2.05i to fix a couple of issues.

3.52
1. Added human voice/Answering machine/Fax detection feature for auto dialer task.

3.51
1. Enhancement on conference room.
Added password protection for conference room.
Added disconnecting last call for conference room.


3.50
1. Fixed an issue with Andriod SIP phone.
2. Added greeting option for conference room member join and leave
3. Added more log for ACD select.
4. Added CALLID column for status_channel table and populated this value into callid column.
5. Populated call's RecordAudioFile into status_channel's RecordFile column.
6. Added call parking log and call pickup log into IVRKeys column of cdr_table. Also call parking and call pickup calls are set same unique call id as original calls.
7. Enhanced logic of outbound call percentage to prevent outbound calling failed issue.

3.49
1. Added always forwarding option for extension.
2. Used latest SDK API dll, which supposed to address two issues:
a. Logging crashing issue.
b. Reinvite codec priority choosing issue.

3.48a
1. Fixed a GUI bug to set voicemailbox of second extension.
2. Fixed a ACD transfercall issue caused by ACDMusicOnHold error.

3.48
1. Added transferred tag for cdr_pbx, cdr_acd, and cdr_exten
This tag is to mark if the call has been transferred.

2. Added Skill Level for acd agent, so can route calls based on skill level of acd agent.

3. Added IVRKeys column for cdr_pbx to record what clients have pressed on IVR menu.

4. Added TaskID for table auto_dialer_jobs and auto_dialer_done.

5. Fixed a bug for outbound calls from auto_dialer forwarded to ACD huntgroup.


3.47
1. Integrated with latest SDK DLL, which solved a stability issue.


3.46
1. Added support in SIP account, to allow a SIP account map to extension's registered IP address and port.
This is useful for some Gateways which can register on IPPBX as extension.
This allows you put gateway on a dynamical IP address environment, or behind NAT.

2. DB updates to 3.46

3. Fixed a call cancel issue for AudioCodes GW connected to Nortel PBX.
Added option in System settings, to allow enable or disable "getting remote contact address from ring or session progres".

4. Fixed a transfer issue when ringing timeout on destination extension.

