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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12287 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r11946 | russell | 2006-03-05 21:32:35 -0500 (Sun, 05 Mar 2006) | 2 lines

fix a typo in the description of the ringtimeout option

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 usegmtime, log date/time in GMT
 loguniqueid log uniqueid
 loguserfield log user field
 patch provided by bug 5015


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https://origsvn.digium.com/svn/asterisk/branches/1.2

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r10736 | kpfleming | 2006-02-22 11:32:15 -0600 (Wed, 22 Feb 2006) | 2 lines

add comment warning people about trying to use hostnames/IPs in the sample config

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10737 65c4cc65-6c06-0410-ace0-fbb531ad65f3

config to other asterisk samples , bug note 6530


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10559 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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asterisk standard, changed it so it can take frames
 of sizes other than 20ms, allowed for the app to reload
 properly, and finalyl changed sample to general section
 as to follow standards.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9960 65c4cc65-6c06-0410-ace0-fbb531ad65f3

fixes.  Thanks to all those involved.


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- Moving the timezone section to below the documentation section

Example written and contributed by Henrik Olsen, Astricon Training student.
Thanks!



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9292 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9186 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* 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

- Moving files from / to /doc or /configs
- Renaming some documentation files

Thank you for the initiative, manxpower!


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9046 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- Add some extra headers and reference to other doc/ files for realtime


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From now on, apply maxexpiry and minexpiry to all subscriptions. Thanks to fourcheeze in the IRC channel
for pointing this out.


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cleaning patch 6267


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- Add callingpres to sip.conf
- Add reference to README.callingpres from zapata.conf


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with other channels.


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monitor app from manager and in the call through
features.conf bug 5395 for the patch


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8063 65c4cc65-6c06-0410-ace0-fbb531ad65f3

autopause.  thanks twisted from 5577


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other part.


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as multiple agents at same time. bug 6046 thanks
ppyy


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fixes.


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