Christian Richter
4be235a974
added bearer capability reject support. we send release instead of disconnect in case we have no real channel yet. added support for Restarting channels added support for sending complete decoding. changed some log levels.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01 12:51:41 +00:00
Christian Richter
8122c35675
fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 07:58:52 +00:00
Christian Richter
19d46333bf
added callcounters for incoming and outgoing calls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 15:02:03 +00:00
Christian Richter
efccf89eae
Added option far_alerting. This option makes it possible to generate a Ringing on other channels if they feel that they should have inband ringing, but there is non in reality. I need this due to the fact that asterisk has not the possibility to transmit progress indicators thus chan_sip and others do not know wether they should generate a Rining tone themselves if they receive AST_CONTROL_RINGING..
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-05 16:38:15 +00:00
Russell Bryant
c38fbd246e
note that group assignments must be from 0 to 63 (issue #7048 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-28 16:42:42 +00:00
Christian Richter
0b6bd0073b
put the default misdn.trace to /var/log/asterisk/misdn.log for better integration of existing log structure
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-27 08:23:53 +00:00
Christian Richter
52eb1ad9d1
removed dynamic switching from transparent to hdlc mode. Instead we've got a config option hdlc=yes now which enables the hdlc controller for a data call
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-20 18:04:05 +00:00
Christian Richter
a0800bd179
these traceing option do not exist anymore
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-20 10:00:34 +00:00
Christian Richter
8e7dd52695
added option to change the connected party number dialplan (ton)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-09 18:01:27 +00:00
Christian Richter
21735de56d
added a bit more detailed description for the echotraining parameter, also changed the default from 1 to 2000. The default for the upper_threshold is now 0
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-07 11:08:09 +00:00
Christian Richter
bd9c89a710
better default values for jitterbuffer in code and config
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-28 11:46:55 +00:00
Christian Richter
afaf8e4c04
adde incoming_early_audio option, to avoid sending tone indications to the remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-15 19:51:33 +00:00
Christian Richter
f6bd1b8559
added pmp_l1_check option, to avoid l1 checking for group calls on PMP ports
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-15 19:32:45 +00:00
Christian Richter
b42dd639ee
default values of jitterbuffer and jitterbuffer_upper_threshold should be > 0, this fixes the tv_fix warnings, because we use ast_read to transmit frames to asterisk in jitterbuffer mode, instead of queueing the audio data with ast_queue_frame.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-07 13:34:59 +00:00
Christian Richter
7133d1b006
* removed unnecessary struct elements and functions
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* fixed "RETRIEVE does not work" bug
* fixed some NT Mode bugs
* removed some // comments
* added configureable jitterbuffer
* removed own tone-generator, and use asterisks instead, to support
asterisks indications
* added more support for hw-bridging, we bridge now every possible call
* fixed some hdlc mode issues, with a patch for chan_zap we can make
data calls between chan_zap and chan_misdn now
* completely reworked the config engine, works like a charm now
* fixed SetCallerPres - bug
* added Progress and Proceeding passing
* optimized Ringing Indication handling
* added full ast_send_text support (you can setup nice menus with the dialplan
now)
* added support to read /etc/misdn-init.conf to clarify the NT+PTP Problem
* we compile now channels/misdn if mISDNuser is installed systemwide
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-02 21:15:34 +00:00
Christian Richter
d37857c208
updated the documentation and the sample config to meet the present
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-12 22:26:35 +00:00
Kevin P. Fleming
2c65582b66
remove extraneous svn:executable properties
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29 18:24:39 +00:00
Kevin P. Fleming
986a8ca089
issue #5566
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-01 22:04:14 +00:00
Kevin P. Fleming
0ac988acaa
add experimental mISDN channel driver (issue #4077 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-10-31 22:51:12 +00:00