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  - "tuttter" changed to "stutter" ;-)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32252 65c4cc65-6c06-0410-ace0-fbb531ad65f3

   Thanks to everyone involved in working with this!


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  and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
  the sip, zap, and skinny channel drivers, as copying the same global
  configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257, north, with additional updates and modifications)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31324 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
r31321 | kpfleming | 2006-06-01 07:41:47 -0500 (Thu, 01 Jun 2006) | 2 lines

remove a sample entry that never should have been added (code to support it was not merged)

........


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and fixes bug with pruneregister


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31275 65c4cc65-6c06-0410-ace0-fbb531ad65f3

jitterbuffer!


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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)

Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!


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

........
r30239 | file | 2006-05-25 12:27:44 -0300 (Thu, 25 May 2006) | 2 lines

Get rid of an incorrect SIP dial string in the sample extensions.conf - I even tried variations... no go (issue #7222 reported by arkadia)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30240 65c4cc65-6c06-0410-ace0-fbb531ad65f3

fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29938 65c4cc65-6c06-0410-ace0-fbb531ad65f3

use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29903 65c4cc65-6c06-0410-ace0-fbb531ad65f3

patch 0006763 with minor changes.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29553 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 This is part 2/2 of the patches for #7090. Adds one-step call parking to /trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340.



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 - with contributions from miconda, jcollie, and sb
 - branch maintained by oej
Thanks everyone!


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29094 65c4cc65-6c06-0410-ace0-fbb531ad65f3

add another media path reinvite 'flavor', where we will only redirect our media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
also, documented the 'canreinvite=update' option in the sample config file


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28215 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allow contexts in regexten so that extensions can be added to multiple contexts when peer registers (issue #6869 reported by and created by Marquis)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28168 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add distinctive ring detection with Caller ID for Australia, New Zealand, and other countries. (issue #3596 reported by deon patch by dbowerman with minor mods by moi)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27812 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- Optionally send systemname in manager (cool when you have a manager proxy)
- Use systemname in CLI prompt


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goes through the existing path.


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 Integrate the MixMonitor functionality (introduced in 1.2) as an option for recording queue member conversations with callers. #7084



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25076 65c4cc65-6c06-0410-ace0-fbb531ad65f3

is still set (issue #6120, khaefner)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25057 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 Allow for the execution of an AGI to the caller's channel right before they get bridged with the queue member that is going to take their call. Add the option to set a MEMBERINTERFACE variable on the caller's channel that will contain the interface of the queue member that is going to/did take the call. #6843 



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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..

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24879 65c4cc65-6c06-0410-ace0-fbb531ad65f3

 Fix autofill behavior in app_queue and document it's functionality in queues.conf.sample and UPGRADE.txt



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

- add information about subscriptions and realtime dial plans in sip.conf.sample


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put the default misdn.trace to /var/log/asterisk/misdn.log for better integration of existing log structure

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22795 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merge Steve Murphy's (murf) complete re-implementation of AEL, which is now no longer considered experimental :-)


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