166 Commits (cc21007de174e228efa1a349e0cada08be7537ca)

Author	SHA1	Message	Date
Kevin P. Fleming	e9d22f802e	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/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	347665503e	T.38 change note is not necessary in this branch ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	0a6e06c7ff	Rework of T.38 negotiation and UDPTL API to address interoperability problems ... Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	aa379bb741	Document the "flag" field in the voicemessages table. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ff4bfb966a	Merged revisions 204556 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	34d0143955	Recorded merge of revisions 204469 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c511a26749	Move Asterisk-addons modules into the main Asterisk source tree. ... Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	b7feca3685	Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. ... There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. ... This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5e256effa7	Update UPGRADE.txt and CHANGES for 1.6.3 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a80ef0ca1e	Add a note about the ordering of entries in sip.conf in 1.6.1. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. ... This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	9a7c28cd5a	we can now build with -Wformat=2, which found a couple of real bugs ... because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	17ed84ff07	Make the Join event from app_queue use CallerIDNum insead of CallerID for ... indicating the callerid number just like the rest of asterisk. (closes issue #13883) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	9789c66375	as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	773eda05d0	move relevant entries into UPGRADE.txt and resync UPGRADE-1.6.txt with previous branches ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157739 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	81a16aa982	make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines ... also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	cf6c66de65	Fix some refcounting in app_queue.c and change the ... hashing used by app_queue.c to be case-insensitive. This is accomplished by adding a new case-insensitive hashing function. This was necessary to prevent bad refcount errors (and potential crashes) which would occur due to the fact that queues were initially read from the config file in a case-sensitive manner. Then, when a user issued a CLI command or manager action, we allowed for case-insensitive input and used that input to directly try to find the queue in the hash table. The result was either that we could not find a queue that was input or worse, we would end up hashing to a completely bogus value based on the input. This commit resolves the problem presented in issue #13703. However, that issue was reported against 1.6.0. Since this fix introduces a behavior change, I am electing to not place this same fix in to the 1.6.0 or 1.6.1 branches, and instead will opt for a change which does not change behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	59d9255977	Break up skinny.conf into seperate sections for ... devices and lines. (closes issue #13412) Reported by: wedhorn Patches: config-restruct-v4.diff uploaded by wedhorn (license 30) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9335b3ad34	Allow people to select the old console behavior of white text on a black ... background, by using the startup flag '-B'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	47ea5a01b4	Merged revisions 137530 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug 2008) | 1 line add document describing what users will need to be aware of when upgrading to this version and using DAHDI ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	9c7099faae	Log the userfield CDR variable like the other CDR backends, assuming the ... column is actually there. If it's not, we still log everything else as before. (closes issue #13281) Reported by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ff101d0b07	Add '+=' append operator to configuration files. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	75d38f6024	Change SendImage() to output a more consistent status variable. ... (closes issue #13134) Reported by: eliel Patches: app_image.c.patch uploaded by eliel (license 64) UPGRADE.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1517710d7e	Change several 'core' commands to be 'dialplan' commands (with appropriate ... deprecation, of course) (closes issue #13016) Reported by: caio1982 Patches: dialplan_globals6.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	19830f3359	Merge in changes from my cdr-tds-conversion branch. This changes the internal ... implementation from using the volatile libtds, to using the db-lib front end. The unintended side effect of this is that we support (at least) versions 0.62 through 0.82 of the FreeTDS distribution without any #ifdef ugliness. (closes issue #12844) Reported by: jcollie git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Sean Bright	00f74ac24c	Update CHANGES and UPGRADE.txt per kpfleming's mail to #asterisk-dev. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	b2ef18dab4	Add some more IAX2-specific information about the channel to the CHANNEL() ... function and begin the transition from SIPCHANINFO() to just using CHANNEL(). (closes issue #12856) Reported by: mostyn Patches: iax_and_sip_channel_info.patch uploaded by mostyn (license 398) (with some additional cleanup by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	99c2f1c9f7	Expand CDR uniqueid field to 150 chars, to account for maximum systemname. ... (Closes issue #12831) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	77f9f76bc2	Add info on the [compat] section of asterisk.conf. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	316e334751	Change space-zero to now evaluate to false, as is expected by a great many. ... (Inspired by a post on the -users list) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	a92223a240	Modify externnotify to take the number of urgent voicemails as a final argument instead ... of the string "Urgent" (closes issue #12660) Reported by: jaroth Patches: externnotify.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7daebcd610	Adding support for "urgent" voicemail messages. Messages which are ... marked "urgent" are considered to be higher priority than other messages and so they will be played before any other messages in a user's mailbox. There are two ways to leave an urgent message. 1. send the 'U' option to VoiceMail(). 2. Set review=yes in voicemail.conf. This will give instructions for a caller to mark a message as urgent after the message has been recorded. I have tested that this works correctly with file and ODBC storage, and James Rothenberger (who wrote initial support for this feature) has tested its use with IMAP storage. (closes issue #11817) Reported by: jaroth Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent Tested by: putnopvut, jaroth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	2e6537c5db	Note change for ExecIf syntax (caught by jmls on IRC) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	705ba4304d	clarify wording ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	2c52c79e10	Clarify the deprecation notice about Macro() to note that it will not be removed ... for the sake of backwards compatibility, since it is a non-trivial task to convert existing large dialplans that depend on Macro() to use GoSub(), instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	8b1cb3ce53	Make app_directory dependent on app_voicemail. This is because the function ... which says the person's name is handled inside app_voicemail now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	769abc6053	Merged revisions 111126 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines update UPGRADE notes to document usage of the script ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111127 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	caf7b47b69	Merged revisions 110962 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines add note that the user will need to enable codec_ilbc to get it to build ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	789831ef9a	Merged revisions 110880 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	8a411ccf83	Create a centralized configuration option for silencethreshold ... (closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f8412a637d	Deprecate the "stripmsd" option in favor of dialplan substring variable syntax. ... (closes issue #12060) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	f92a3e119e	Move Originate to a separate privilege and require the additional System privilege to call out to a subshell. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	3e0f3915a5	Add CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan application. This will either be set to NOCHANNEL if the given channel was not found or SUCCESS if it worked. ... (closes issue #11553) Reported by: johan Patches: UPGRADE.txt.channelredirect.patch uploaded by johan (license 334) CHANGES.channelredirect.patch uploaded by johan (license 334) app_channelredirect-20080219.patch uploaded by johan (license 334) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	44810652d6	Change the queue holdtime announcement to happen at any interval (not just greater than two minutes). Remove ... the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using '1' as a queue-round-seconds value is no longer valid. (closes issue #9736) Reported by: caio1982 Patches: queue_announce5.diff uploaded by caio1982 (license 22) Tested by: caio1982, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	8c3cf89933	1. Deprecate SetMusicOnHold and WaitMusicOnHold. ... 2. Add a duration parameter to MusicOnHold (closes issue #11904) Reported by: dimas Patches: v2-moh.patch uploaded by dimas (license 88) Tested by: dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	18347a73ff	At the request of ManxPower, include the UPGRADE.txt from 1.2 and 1.4, as well. ... This way, if people need to go back and review what was deprecated in previous major releases, it is readily available to them. Thanks for the suggestion! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	10f6450da7	Add a note about changing modules.conf since another console channel driver is ... now present that can not be used at the same time as chan_alsa or chan_oss. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	17afebc1a6	HUGE improvements to QoS/CoS handling by IgorG ... - Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago