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1) Added the unistim.conf variable dtmf_duration which can select the DTMF playback duration from 0ms to 150ms (0 is off and is the new default)
2) Enabled the transmission of month names, which are sent with the date and changed the dateformat variable to accept the values 0-3 as per the UNISTIM standard (2 & 3 match the previous 1 & 2 formats).
3) Enabled the "Mute" packet so muting microphone works as expected and microphone muted for all calls while LED light on
4) Changed Duree to Timer on i2004 display
(closes issue ASTERISK-23592)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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hardcoded dial timeout constant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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channel driver operation and add additional functionality:
* Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls.
* Added ability for translation on-screen menu to multiple languages. Tested on Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone
* Other described in CHANGES file
Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa.
Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E.
(closes issue ASTERISK-16890)
Review: https://reviewboard.asterisk.org/r/1243/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Adding links to http(s)://wiki.asterisk.org
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #17600)
Reported by: minaguib
Patches:
ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
Tested by: minaguib
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Let's try that again, this time removing trailing whitespace and not leading
whitespace. I can't believe no one noticed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11168)
Reported by: IgorG
Patches:
unistimjb-88863-1.patch uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones. The following models have been confirmed
to work: i2002, i2004 and i2050.
(closes issue #8864)
Reported by: c_hans
Patches:
chan_unistim.patch uploaded by c (license 304)
ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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