v0.1-beta.3

Bug fixes over Beta 2. Nothing planned to change for this version.

• Manual calls are now fully functional

Is functional and stable, some CallLog bits to clean up but nothing that affects daily use.

v0.1-beta.2

Beta marches onwards

• Bug fixes for phone ringing, update to this or you'll end up with left over panels due to old ringing code
• Added chat alert noise for studio use
• Added manual call button to add call information from a caller outside of the system (e.g. POTS)

Need to work in a way of changing the callers number on a manual call. Would also like to refactor parts of the XMPP messaging code to make it more understandable however this is quite an internal change and would require lots of testing.

v0.1-Beta.1

This has been running in a production setting for quite some time so its probably worth attaching a beta tag to the release.

New Features for this release:

• Setting: playRinging - Audible alert for when calls are ringing, set it to true if you want a machine/user/global to play ringing noises
• Some tweaks for newer versions of Elastix
• Minor tweak to call handling code that looks at trunk CIDs. Some SIP providers provide the full trunk number (e.g. 0044123456789) on incoming calls which is different on an older PRI based ISDN line I tested with

There is still a lot to tidy up but the basic mechanics all work perfectly fine. I recommend closing the program when you've finished with it rather than letting PCs hibernate as you can lose XMPP connection but the program thinks it is still connected so you get no errors.

Also be aware if you have very high ping remote offices (approaching 1,000ms) you can get XMPP drop outs. I'm pretty sure this is due to the VPN connection breaking down rather than a program "feature".

v0.1-alpha.4

WARNING

Update your database Schema: All previous versions including this one only expected Asterisk channels as TIMESTAMP+3 Digits. This means that the program will only work for the first 1,000 calls depending on your Asterisk implementation.

Run the following commands on your callhistory and conversations table to solve this problem:

ALTER TABLE `callhistory` MODIFY `callchannel` VARCHAR(80);
ALTER TABLE `conversations` MODIFY `channel` VARCHAR(80);

Note: This will increase the table storage costs over the previous DECIMAL values but it is required to deal with Asterisk channels. I've checked this on several asterisk implementations and they all seem to use VARCHAR(80) when storing channels in the DB so this should be sufficient to prevent this problem happening again.

Old release info for archive purposes

Alpha 4 release for testing of Asterisk Phone both Server and Client.

This will probably be the last release before Beta with only the call log spurious records and stealing outbound to fix.

Improvements since Alpha 3:

• Various Bug Fixes
• Outgoing calls are now tracked correctly with all the channel swapping weirdness that that entails (exception being stealing an outgoing call)

Upgrading from Alpha 2 and 3 [Optional]:

Nothing to do just run it.

Upgrading from Alpha 1 [Optional]:

I changed the database to increase the column size for the phone number variable. This won't affect the running of the program as values will just be truncated if they are too big. However if you want to stay up to date just run an alter command:

ALTER TABLE phonenumbers MODIFY phone_number VARCHAR(20);
ALTER TABLE callhistory MODIFY phonenumber VARCHAR(20);

Known Issues:

Internal logic for caller id relates to the UK number formats which is not applicable for quite a few places:

• < 6 Digits = Internal
• < 8 Digits = Local
• < 12 Digits = National
• 12+ Digits = International

Outbound calls to internal numbers won't keep call info once they land in a queue if you use dial prefixes e.g. 9 for outside line. This is due to programmer idiocy always removing the dial prefix even when its internal. Fixed in master.

If you steal an outbound call before it enters a queue the info won't transfer properly as the channel switch code relied on the normal chain of call -> queue -> operator.

Unfinished features:

• Various hard coded field info means that even though the form can be flexibly built using the database, the records that do things behind the scenes are mostly hard coded. Needed to do this to preserve sanity and release something before the Sun explodes.

v0.1-alpha.3

Alpha 3 release for testing of Asterisk Phone both Server and Client.

Improvements since Alpha 2:

• Various Bug Fixes
• Can now Change a Person either due to phone sharing or with held number
• Everything Call Log related now updates on the fly

Upgrading from Alpha 2 [Optional]:

Nothing to do just run it.

Upgrading from Alpha 1 [Optional]:

I changed the database to increase the column size for the phone number variable. This won't affect the running of the program as values will just be truncated if they are too big. However if you want to stay up to date just run an alter command:

ALTER TABLE phonenumbers MODIFY phone_number VARCHAR(20);
ALTER TABLE callhistory MODIFY phonenumber VARCHAR(20);

Known Issues:

Internal logic for caller id relates to the UK number formats which is not applicable for quite a few places:

• < 6 Digits = Internal
• < 8 Digits = Local
• < 12 Digits = National
• 12+ Digits = International

Unfinished features:

• Various hard coded field info means that even though the form can be flexibly built using the database, the records that do things behind the scenes are mostly hard coded. Needed to do this to preserve sanity and release something before the Sun explodes.

v0.1-alpha.2

Alpha 2 release for testing of Asterisk Phone both Server and Client.

Upgrading from Alpha 1 [Optional]:

I changed the database to increase the column size for the phone number variable. This won't affect the running of the program as values will just be truncated if they are too big. However if you want to stay up to date just run an alter command:

ALTER TABLE phonenumbers MODIFY phone_number VARCHAR(20);
ALTER TABLE callhistory MODIFY phonenumber VARCHAR(20);

Known Issues:

Internal logic for caller id relates to the UK number formats which is not applicable for quite a few places:

• < 6 Digits = Internal
• < 8 Digits = Local
• < 12 Digits = National
• 12+ Digits = International

Unfinished features:

• Call Log only updates on load, need to finish on the fly updates
• No way of changing a person attached to a phone number if more than one person uses the same phone (or people withheld their numbers)

v0.1-alpha.1

Initial release for testing of Asterisk Phone both Server and Client.

Known Issues:

• MySQL connections are not reopened if they get closed by the MySQL Server
• Call Locations for National/Local numbers are hard coded in the records.strings.properties file so change these if you don't live in the UK for the Clients.
• Clients are writing initial phone number info on first time callers, could cause race conditions with multiple clients, need to move this to Server side.
• First time callers won't save properly but will save the second time they call

Unfinished features:

• Call Log only updates on load, need to finish on the fly updates
• No way of changing a person attached to a phone number if more than one person uses the same phone (or people withheld their numbers)