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

Kris Boutilier reviewboard at asterisk.org
Fri Dec 9 12:37:40 CST 2011


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1243/#review4981
-----------------------------------------------------------


What additional testing is required before this can be merged?

- Kris


On Sept. 12, 2011, 4:33 a.m., IgorG wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1243/
> -----------------------------------------------------------
> 
> (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
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20111209/98fdd10b/attachment-0001.htm>


More information about the asterisk-dev mailing list