asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin P. Fleming	bbf4a08fe7	Merged revisions 210190 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210190 \| kpfleming \| 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) \| 11 lines Rename 'canreinvite' option to 'directmedia', with backwards compatibility. It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@210191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	417c6dcb1d	Update res_phoneprov to default to setting the SERVER variable to the IP the HTTP request for the config came in on and the SERVER_PORT to the bindport setting in sip.conf. I've left in the ability to override these options, because I can't always guess how someone might decide to do something weird with what is available to them--although needing to is pretty unlikely. Documentation was updated to reflect preference for not setting serveraddr, serveriface, or serverport. Tested on Linux and OS X. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	3570ad103d	Added a new module, res_phoneprov, which allows auto-provisioning of phones based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

Author

SHA1

Message

Date

Kevin P. Fleming

bbf4a08fe7

Merged revisions 210190 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines
  
  Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
  
  It is clear from multiple mailing list, forum, wiki and other sorts of posts
  that users don't really understand the effects that the 'canreinvite' config
  option actually has, and that in some cases they think that setting it to 'no'
  will actually cause various other features (T.38, MOH, etc.) to not work properly,
  when in fact this is not the case. This patch changes the proper name of the
  option to what it should have been from the beginning ('directmedia'), but
  preserves backwards compatibility for existing configurations.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@210191 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Terry Wilson

417c6dcb1d

Update res_phoneprov to default to setting the SERVER variable to the IP

the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf.  I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.

Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport.  Tested on Linux and OS X.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Terry Wilson

3570ad103d

Added a new module, res_phoneprov, which allows auto-provisioning of phones

based on configuration templates that use Asterisk dialplan function and
variable substitution.  It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3

3 Commits (c9a23f6d78da2bf98d3784dc2ca74194e212800d)