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<span class="Apple-style-span" style="font-family: Verdana, Arial, Helvetica, sans-serif; font-size: 13px; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px; ">Hi All,</span></div><div><br></div>
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. Please visit mantis to find out patch for current svn trunk: <meta http-equiv="content-type" content="text/html; charset=utf-8"><a href="https://issues.asterisk.org/view.php?id=18229">https://issues.asterisk.org/view.php?id=18229</a><br>
<br>New unistim.conf options:<br>- Added "debug" global option in unistim.conf, that enable debug when module loaded<br>- Added "sharpdial" option, enable sending call whet # key pressed<br><br>New features:<br>
- ability for changing display language (tested on Russian language). Use .po files in encoding, able to display<br> ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. For selecting language can be used option "language" in<br>
unistim.conf or screen menu.<br>- Support for multilines<br>- Support for holding multiple lines<br>- More fixes for display on i2002 phone<br>- Configurable keys for sending and received history<br>- Menu for selecting codec, contrast (not yet completed) or display language<br>
- Show clock at first line of idle phone <br>- Add ability for pick up call <br>- Pick up call by using on-screen soft key<br>- Change displaying list of received or send calls (callerid, time and caller name on different screens, listed by lef-right keys)<br>
<br>Changes:<br>- Changed entering on screen phone number, so any number of digits can be entered<br>- rtp_port now used start rtp port<br>- list of dial tone frequecies now loaded from indications.conf and not hardcoded<br>
- Key with globe icon how calls menu and not directly codec selection<br><br>Fixes:<br>- <a href="https://issues.asterisk.org/view.php?id=17406" title="[closed] Speakerphone LED Update Fail after call termination.">0017406</a> Correct updating LED when switching between speekerphone and handset or hanging up<br>
- <a href="https://issues.asterisk.org/view.php?id=17327" title="[closed] Random crash in chan_unistim">0017327</a> Multiple crashes when using phone <br>- <a href="https://issues.asterisk.org/view.php?id=16867" title="[closed] Phone generated dial tone not switched off during call">0016867</a> Fixed playing dialtone in some scenarious when conversation already started<br>
- Fixed dispalying on-screen information when using Redial softkey (DN number and timer displayed). <br>- Not sending short ring in case of call forward enabled on phone</span><br clear="all"><br>-- <br>Regards,<br>Igor Goncharovsky<br>
Blog: <a href="http://igorg.ru" target="_blank">http://igorg.ru</a><br>