[asterisk-dev] [Code Review]: Update to chan_unistinm functionality

[Matt Jordan]

> On Dec. 9, 2011, 12:37 p.m., Kris Boutilier wrote:
> > What additional testing is required before this can be merged?

Looking at the comment traffic flow on the Asterisk issue, it looks as if the most recent diff isn't the most recent attached to that issue.  When I skimmed through this diff there are a few coding guideline fixes that could be made, but - based on the comments - it looks like the community has vetted this patch.  I'd like to see the latest and greatest patch uploaded here and one final pass through made by the community members who have tested it to verify that they're happy with it.


- Matt


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On Sept. 12, 2011, 4:33 a.m., IgorG wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1243/
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> 
> (Updated Sept. 12, 2011, 4:33 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> New unistim.conf options:
> - Added "debug" global option in unistim.conf, that enable debug when module loaded
> - Added "sharpdial" option, enable sending call whet # key pressed
> 
> New features:
> - ability for changing display language (tested on Russian language). Use .po files in encoding, able to display
>   ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. For selecting language can be used option "language" in
>   unistim.conf or screen menu.
> - Support for multilines
> - Support for holding multiple lines
> - More fixes for display on i2002 phone
> - Configurable keys for sending and received history
> - Menu for selecting codec, contrast (not yet completed) or display language
> - Show clock at first line of idle phone
> - Add ability for pick up call
> - Pick up call by using on-screen soft key
> - Change displaying list of received or send calls (callerid, time and caller name on different screens, listed by lef-right keys)
> 
> Changes:
> - Changed entering on screen phone number, so any number of digits can be entered
> - rtp_port now used start rtp port
> - list of dial tone frequecies now loaded from indications.conf and not hardcoded
> - Key with globe icon how calls menu and not directly codec selection
> 
> Fixes:
> - 0017406 Correct updating LED when switching between speekerphone and handset or hanging up
> - 0017327 Multiple crashes when using phone
> - 0016867 Fixed playing dialtone in some scenarious when conversation already started
> - Fixed dispalying on-screen information when using Redial softkey (DN number and timer displayed).
> - Not sending short ring in case of call forward enabled on phone
> 
> 
> This addresses bug 18229.
>     https://issues.asterisk.org/jira/browse/18229
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/chan_unistim.c 333499 
> 
> Diff: https://reviewboard.asterisk.org/r/1243/diff
> 
> 
> Testing
> -------
> 
> Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa. 
> 
> 
> Thanks,
> 
> IgorG
> 
>

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