[asterisk-bugs] [Asterisk 0018229]: [patch] Update for chan_unistim fuctionality

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18229 
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Reported By:                IgorG
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18229
Category:                   Channels/chan_unistim
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
JIRA:                       SWP-2488 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 293198 
Request Review:              
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Date Submitted:             2010-10-28 23:55 CDT
Last Modified:              2011-04-09 13:43 CDT
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Summary:                    [patch] Update for chan_unistim fuctionality
Description: 
During last three month I have worked on improving functionality of Nortel
phones working with asterisk to replace existing Nortel station by
asterisk. Many improvments done, listed below. I have only i2002 phone and
unable to test if new version of channel correctly works with i2204 phone.
If anyone can test it and report issues, it would be great 

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:
- https://issues.asterisk.org/view.php?id=17406 Correct updating LED when
switching between speekerphone and
handset or hanging up
- https://issues.asterisk.org/view.php?id=17327 Multiple crashes when using
phone 
- https://issues.asterisk.org/view.php?id=16867 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

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Relationships       ID      Summary
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parent of           0017406 Speakerphone LED Update Fail after call...
parent of           0017327 Random crash in chan_unistim
parent of           0016867 Phone generated dial tone not switched ...
related to          0018540 Problem with unistim on Asterisk 1.8.1.1
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---------------------------------------------------------------------- 
 (0133564) idarwin (reporter) - 2011-04-09 13:43
 https://issues.asterisk.org/view.php?id=18229#c133564 
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Not quite, IgorG. In more detail;
(SIP Phone, i2004) both connected to local Asterisk
Local asterisk connected to PSTN line (via an ATA for historical reasons)
PSTN line (Bell Canada) connected to:
   a) Rented DID connected to remote Asterisk via IAX2 from VOISP, and
   b) Voice mail provided by Bell Canada
In both cases, A & B, SIP phone works with IVR, Unistim phone does not
work with IVR. In case B, PSTN-provided Voice Mail IVR echoes back some
random subset of the digits entered. So, I set up the obvious test with
Read() and SayNumber(), on both Asterixes...

Oops! On local, I was surprised to see that *both phones pass*, but I'm
not sure that is a 'real' test - does it really rely on DTMF decoding on a
local call? If so, the problem is at my end, not in chan_unistim! On
remote, I'd been lucky, the SIP phone gets occasional mistakes too when
running this test.

I know - let's blame the ATA. I'll go investigate there - looks like the
problem is outside your module.  Sorry for the noise. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-04-09 13:43 idarwin        Note Added: 0133564                          
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