asterisk

Commit Graph

Author	SHA1	Message	Date
Leif Madsen	74bf257e39	Merged revisions 258351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258351 \| lmadsen \| 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010) \| 20 lines Add ability to generate ASCII documentation from the TeX files. These changes add the ability to run 'make asterisk.txt' just like the existing 'make asterisk.pdf' commands to generate a text document from the TeX files we have in the doc/tex/ directory. I've also updated a few of the .tex files because they weren't properly escaping certain characters so they would show up as Unicode characters (like [U+021C]). Made changes to the configure scripts so it would detect the catdvi program which is required to convert the .dvi file generated by latex. I've also added a few lines to the build_tools/prep_tarball script so that the text documentation gets generated and added to future tarballs of Asterisk releases. (closes issue #17220) Reported by: lmadsen Patches: asterisk.txt.patch uploaded by lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger (license 224) Tested by: lmadsen, pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@258384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
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	17 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

4 Commits (cef5e9d4340f94de8801aaecfb004a080df8a68f)