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- Add some comments


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as well as being more efficient, patch from jcollie's
base64 branch


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potential problems


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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)


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(issue #7258, tardieu)


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- change references to the "scx" jitterbuffer to be called "fixed" and change
  references to the "stevek" jitterbuffer to be called "adaptive", instead


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

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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 adds check into jabberstatus so it cant segfault.


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and fixes bug with pruneregister


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- improved support of attended transfers (REFER with replaces)
- support of INVITE/replaces in the context of a transfer
- improved support of blind transfers (REFER)

Thanks to Voop, Nuvio and Foniris for sponsoring this work.


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jitterbuffer!


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different way that actually works


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

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r31194 | russell | 2006-05-31 19:50:00 -0400 (Wed, 31 May 2006) | 3 lines

if the connection to a FastAGI server fails because of a timeout, log a more
informative log message

........


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

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r31161 | kpfleming | 2006-05-31 17:26:38 -0500 (Wed, 31 May 2006) | 2 lines

silence a warning message that is not a warning

........


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These used to be wrapped in a #ifdef


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tech structures indicate that they create jitter


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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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OKAY - Only replace our IP address in the outgoing SIP messages if the actual source address is within the specified local network. (issue #7051 reported by Luke-Jr with mods by me)


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cleanups for commit from issue #5657... set a cause code for a rejected forward request, and actually set tmp->chan to NULL when we reject the forward request


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 Add an option to app_dial, 'i', to instruct the application ignore any requests from peers to forward calls elsewhere. #5657 (johnlange w/some minor mods)



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