aboutsummaryrefslogtreecommitdiffstats
path: root/include
AgeCommit message (Collapse)AuthorFilesLines
2008-11-01Merge changes from team/group/appdocsxmlrussell9-6/+220
This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31* Fixed timeout logic in the dialing API as setting timeoutsmmichelson1-2/+2
had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153223 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31Recent CDR fixes moved execution of the 'h' exten into the bridging code, so ↵twilson1-0/+1
variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153181 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30Add a todo for a new timing API implementation that would work for Linux systemsrussell1-0/+3
as of kernel 2.6.25 and glibc 2.8 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152990 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30Fix a bug in AST_SCHED_REPLACE_UNREF(). The reference count of the objectrussell1-2/+1
_must_ be increased before creating the scheduler entry. Otherwise, you create a race condition where the reference count may hit zero and the object can disappear out from under you. This could also would have incorrectly decreased the reference count in the case that the scheduler add failed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152887 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30try to get this committed before the buildbot complains about a broken treekpfleming1-0/+29
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152810 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-29Merged revisions 152535 via svnmerge from murf2-14/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152536 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-19cleaup of the TCP/TLS socket API:kpfleming1-36/+27
1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address git-svn-id: http://svn.digium.com/svn/asterisk/trunk@151101 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-17Merge codec_consistency branch. This should make sample usage much happier.qwell1-0/+115
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150729 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-17Fix the FRACK! warnings in chan_iax2 when POKE/LAGRQ packets are not answered.tilghman1-0/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150580 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-14Merged revisions 149204 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@149205 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-14Add additional memory debugging to several core APIs, and fix several memorytilghman4-0/+45
leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx git-svn-id: http://svn.digium.com/svn/asterisk/trunk@149199 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-14Merge realtime_update2 branch, which adds a new realtime API call namedtilghman2-2/+32
'update2', which permits updates which match across multiple columns, instead of requiring all tables to have a single unique identifier. All of the other API calls with the exception of 'update' already had the ability to match on multiple fields, so it was a missing and very desireable feature that an API call implementing an update should have this, too. This does not change any outward performance of Asterisk, but it should make life easier for application developers who use the RealTime framework. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148570 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-10Don't include logger.h in asterisk.h by default as it is causing problems ↵seanbright1-2/+0
building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148200 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09only include this for OpenBSD. At least FreeBSD is borked when including itmvanbaak1-0/+2
(closes issue #13649) Reported by: ys git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147899 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09(closes issue #13557)murf2-3/+3
Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147807 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-07Allow people to select the old console behavior of white text on a blacktilghman1-0/+3
background, by using the startup flag '-B'. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@147262 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06Update documentation; AST_THREADSTORAGE() in trunk only takes a singletilghman1-3/+2
argument. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146928 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06All ODBC parts can now use either unixodbc or iodbc.mvanbaak1-0/+6
This allows for the ODBC parts to work on OpenBSD as well. 99.99% of the work is done by seanbright (bow, bow) and I actually did nothing but test and yell at him that it still didn't work :) Thanks for helping out ! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146925 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06Similar to r143204, masquerade the channel in the case of Park being called ↵jpeeler1-0/+2
from AGI. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146923 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06Mvanbaak said this was needed to compile on OpenBSD, so put it in the ↵jpeeler1-1/+1
OpenBSD section. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146920 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06make aescrypt.c compile on OpenBSD againmvanbaak1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146807 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-01Add schedule extensions to app_meetme. In addition, the reporter found atilghman1-0/+33
problem within strptime(3), which we are correcting here with ast_strptime(). (closes issue #11040) Reported by: DEA Patches: 20080910__bug11040.diff.txt uploaded by Corydon76 (license 14) Tested by: DEA git-svn-id: http://svn.digium.com/svn/asterisk/trunk@145649 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-27fix bugs caused by r144949 when MALLOC_DEBUG is definedkpfleming1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@144950 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-27Merged revisions 144924-144925 via svnmerge from kpfleming1-7/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines improve header inclusion process in a few small ways: - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled - simplify the usage of some of these headers in the AEL-related stuff in the utils directory ........ r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines fix some minor issues with rev 144924 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@144949 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-25I added a little verbage to hashtab for the hashtab_destroy func.murf1-1/+5
It was pretty sparsely documented. This update fleshes out the pbx_lua module, to add the switch statements to the extensions in the extensions.lua file, as well as removing them when the module is unloaded. Many thanks to Matt Nicholson for his fine contribution! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@144523 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman1-0/+2
when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Merged revisions 142675 via svnmerge from murf1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142676 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-09Minor fix to docosnuffy1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142000 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-26(closes issue #13366)murf1-0/+2
Reported by: erousseau This was a reasonable enhancement request, which was easy to implement. Since it's an enhancement, it could only be applied to trunk. Basically, for accounting where "initiated" seconds are billed for, if the microseconds field on the end time is greater than the microseconds field for the answer time, add one second to the billsec field. The implementation was requested by erousseau, and I've implemented it as requested. I've updated the CHANGES, the cdr.conf.sample, and the .h files accordingly, to accept and set a flag for the corresponding new option. cdr.c adds in the extra second based on the usec fields if the option is set. Tested, seems to be working fine. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140057 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-25Optional light colored background, for those who use black on white terminals.tilghman1-0/+3
(closes issue #13306) Reported by: Corydon76 Patches: 20080814__bug13306__3.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, pkempgen git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139981 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-22Merged revisions 139553 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy (license 35) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139554 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-10Fix this again so we can compile with shadow warnings enabled and IMAP chosenseanbright1-3/+3
in voicemail. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137112 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-09Merged revisions 136946 via svnmerge from tilghman2-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500 (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) | 2 lines Regression fixes for Solaris ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136947 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-08Merged revisions 136726 via svnmerge from murf1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines (closes issue #13236) Reported by: korihor Wow, this one was a challenge! I regrouped and ran a new strategy for setting the ~~MACRO~~ value; I set it once per extension, up near the top. It is only set if there is a switch in the extension. So, I had to put in a chunk of code to detect a switch in the pval tree. I moved the code to insert the set of ~~exten~~ up to the beginning of the gen_prios routine, instead of down in the switch code. I learned that I have to push the detection of the switches down into the code, so everywhere I create a new exten in gen_prios, I make sure to pass onto it the values of the mother_exten first, and the exten next. I had to add a couple fields to the exten struct to accomplish this, in the ael_structs.h file. The checked field makes it so we don't repeat the switch search if it's been done. I also updated the regressions. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136746 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Merged revisions 136541 via svnmerge from kpfleming1-1/+92
https://origsvn.digium.com/svn/asterisk/trunk ........ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136542 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Merge in a few more changes. This time the include/ directory.seanbright2-11/+11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136402 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06Merged revisions 135899 via svnmerge from tilghman2-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) | 4 lines 1) Bugfix for debugging code 2) Reduce compiler warnings for another section of debugging code (Closes issue #13237) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135900 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06Merged revisions 135841,135847,135850 via svnmerge from mmichelson1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines Merging the issue11259 branch. The purpose of this branch was to take into account "burps" which could cause jitterbuffers to misbehave. One such example is if the L option to Dial() were used to inject audio into a bridged conversation at regular intervals. Since the audio here was not passed through the jitterbuffer, it would cause a gap in the jitterbuffer's timestamps which would cause a frames to be dropped for a brief period. Now ast_generic_bridge will empty and reset the jitterbuffer each time it is called. This causes injected audio to be handled properly. ast_generic_bridge also will empty and reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE frame since the change in audio source could negatively affect the jitterbuffer. All of this was made possible by adding a new public API call to the abstract_jb called ast_jb_empty_and_reset. (closes issue #11259) Reported by: plack Tested by: putnopvut ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel that occurred when I was testing for a memory leak ........ r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines Remove properties that should not be here ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135851 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05Merged revisions 135799 via svnmerge from murf1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135821 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05Add '+=' append operator to configuration files.tilghman1-0/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135717 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05datastore inheritance is a channel feature, so move this definition backkpfleming2-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135681 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05make datastore creation and destruction a generic API since it is not really ↵kpfleming3-38/+121
channel related, and add the ability to add/find/remove datastores to manager sessions git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-04HTTP module memory leakstilghman1-1/+5
(closes issue #13230) Reported by: eliel Patches: res_http_post_leak.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135476 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-03Merge in changes that allow Asterisk to be built against the Hoardseanbright1-0/+6
memory allocator. See doc/hoard.txt for more details. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135405 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Oops, wrong definetilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134703 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-25Deprecate *_device_state_* APIs in favor of *_devstate_* APIstilghman1-2/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133860 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-25Modify the main page of the doxygen documentation to link to a new page ↵russell1-0/+43
dedicated to Asterisk licensing information. The licensing page includes the Asterisk license, as well as a (not yet complete) list of 3rd party libraries that may be used, as well as what license we receive them under. Help filling out this list in the format that I have started in doxyref.h would be much appreciated. :) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133575 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-23Merged revisions 133169 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133171 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-23Merged revisions 132872 via svnmerge from kpfleming1-17/+33
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132964 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-29 05:38:37 +0000