asterisk

Commit Graph

Author	SHA1	Message	Date
TransNexus OSP Development	034a79c303	Updated doc for OSP lookup application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Ruggles	1649ae071c	ExternalIVR information for UPGRADE.txt added a paragraph about the fixes and changes to the ExternalIVR application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c207825dc7	Move an entry from CHANGES to UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2a1dce85b4	Move an entry from CHANGES that should be in UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2081809b07	update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by: miki git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	f62d03a8f3	Store the cause code that is returned when trying to create a channel in ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of overwriting the device state in AVAILSTATUS. (closes issue #14426) Reported by: macli git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	6406f39594	DAHDI ISDN channel names will not allow device state to work. (Interim solution.) Since ISDN works like SIP and not analog ports in regard to devices, the device state based on the ISDN channel number could not work. This has not been an issue until the advent of PTMP NT mode. Previously, ISDN lines were used as trunks and did not have to keep track of specific devices. As an interim solution until device states are properly implemented, the channel name is being changed to the following format to use the generic device state support: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will work with the following restrictions: * The number of devices/phones cannot exceed the number of B channels. (i.e., BRI has 2) * Each device/phone can only have one number. No shared MSN's. * The phones/devices probably should not use subaddressing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	20743ec07d	Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	dd0c76a9d3	Move DAHDI/ISDN channel naming note from CHANGES to UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ff555b5e12	Sync up UPGRADE.txt with the 1.6.2 version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	17180120bf	Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	6f8e099b34	Merged revisions 218798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218798 \| russell \| 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) \| 9 lines Remove the IAXy firmware from Asterisk. The firmware can now be found on downloads.digium.com, where the rest of our binary downloads live. This was the last part of our Asterisk tarballs that was considered non-free by Debian. :-) (closes issue #15838) Reported by: paravoid ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	148552de24	Merged revisions 216085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216085 \| russell \| 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) \| 9 lines Merged revisions 216080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216080 \| russell \| 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) \| 2 lines Add a note about IAX2 to UPGRADE.txt. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
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	17 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	17 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	17 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	18 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	18 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

1 2 3 4

179 Commits (2be88e05c043e8abc1b6a3c94c4699859a7adf68)