Update CHANGES, make section about SIP. This might be a good way to handle
other parts of this file too, as it grows. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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CHANGES
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CHANGES
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@ -6,11 +6,6 @@ Changes since Asterisk 1.4-beta was branched:
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prompts within the Voicemail application by changing them in voicemail.conf
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* enable https support for builtin web server.
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See configs/http.conf.sample for details.
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* add a new option, match_auth_username, to sip.conf,
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to improve the matching of incoming requests.
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If set, and the incoming request carries authentication info,
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the username to match in the users list is taken from there
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rather than from the From: field.
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* Argument support for Gosub application
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* MailboxExists converted to dialplan function
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* Ability to set process limits without restarting Asterisk
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@ -65,3 +60,15 @@ Changes since Asterisk 1.4-beta was branched:
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what Asterisk should set as the maximum number of open files when it loads.
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* Added the jittertargetextra configuration option.
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* Added the URI redirect option for the built-in HTTP server
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SIP changes
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-----------
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* The default SIP useragent= identifier now includes the Asterisk version
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* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
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If set, and the incoming request carries authentication info,
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the username to match in the users list is taken from the Digest header
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rather than from the From: field. This feature is considered experimental.
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* The "musiconhold" and "musicclass" settings in sip.conf are now removed,
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since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
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* The "localmask" setting was removed in version 1.2 and the reminder about it
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being removed is now also removed.
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