1847 Commits (33252a15471494ef7160db49fcc85097649c6228)

Author	SHA1	Message	Date
Steve Murphy	ea898dc6c3	Merged revisions 139764 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines This patch reverts the changes made via 139347, and 139635, as users are seeing adverse difference. I will un-close 13251. Back to the drawing board/ concept/ beginning/ whatever! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	de1af295c8	Merged revisions 139635 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 lines I found some problems with the code I committed earlier, when I merged them into trunk, so I'm coming back to clean up. And, in the process, I found an error in the code I added to trunk and 1.6.x, that I'll fix using this patch also. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	2488366a75	Merged revisions 139347 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines (closes issue #13251) Reported by: sergee Tested by: murf THis is a bold move for a static release fix, but I wouldn't have made it if I didn't feel confident (at least a *bit* confident) that it wouldn't mess everyone up. The reasoning goes something like this: 1. We simply cannot do anything with CDR's at the current point (in pbx.c, after the __ast_pbx_run loop). It's way too late to have any affect on the CDRs. The CDR is already posted and gone, and the remnants have been cleared. 2. I was very much afraid that moving the running of the 'h' extension down into the bridge code (where it would be now practical to do it), would result in a lot more calls to the 'h' exten, so I implemented it as another exten under another name, but found, to my pleasant surprise, that there was a 1:1 correspondence to the running of the 'h' exten in the pbx_run loop, and the new spot at the end of the bridge. So, I ifdef'd out the current 'h' loop, and moved it into the bridge code. The only difference I can see is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this is still an important decision point, I can replicate it if there are complaints. To be perfectly honest, the KEEPALIVE situation is not totally clear to me, and how it relates to a post-bridge situation is less clear. I suspect the users will point out everything in total clarity if this steps on anyone's toes! 3. I temporarily swap the bridge_cdr into the channel before running the 'h' exten, which makes it possible for users to edit the cdr before it goes out the door. And, of course, with the endbeforehexten config var set, the users can also get at the billsec/duration vals. After the h exten finishes, the cdr is swapped back and processing continues as normal. Please, all who deal with CDR's, please test this version of Asterisk, and file bug reports as appropriate! ........ I also made a little fix to the app_dial's 'e' option, that is related to my updates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	618b3eb8f8	Merged revisions 139621 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008) | 5 lines (closes issue #13359) Reported by: Laureano Patches: originate_channel_check.patch uploaded by Laureano (license 265) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	ea788c02e7	remove extra comma typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	060f6d5c3d	Add missing unique id to ParkedCallGiveUp and ParkedCallTimeOut ... manager events (closes issue #13358) Reported by: srt Patches: 13358_parking_events.diff uploaded by srt (license 378) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	1b2e0a4365	Merged revisions 139074 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | 12 lines (closes issue #13263) Reported by: brainy Tested by: murf The specialized reset routine is tromping on the flags field of the CDR. I made a change to not reset the DISABLED bit. This should get rid of this problem. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	04795d963f	These changes are in regards to bug 13249, where users are being surprised by the changes made ... to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x installation where a "make samples" was executed, or where they hand-edited the asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher). (this commit does not totally solve 13249, at least not yet) The change involves issueing a single warning while the AEL file is loading, if: 1. app_set is present in the config file, and set to 1.6 or higher. 2. there are double quotes in an assignment statement (eg x = "hi there";) 3. the warning was not already issued. The standalone app, aelparse, does not (yet) issue this warning. I'd have to have it read in the asterisk.conf file, and that's a bit of hassle. I'll add it if users request it, tho. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	642dd656eb	Move Uniqueid to the end of the event for those that rely on the position ... of the name/value pairs, pointed out by snuffy-home on #asterisk-commits. For those of you who rely on the position of name/value pairs in manager events... stop... that is why associative arrays were invented. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	c8c264b9ad	Add Uniqueid header to ParkedCall manager event. ... (closes issue #13323) Reported by: srt Patches: 13323_unique_id_for_parkedcalls_event.diff uploaded by srt (license 378) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	9ff1434641	Add missing colons to RTCPReceived and RTCPSent manager events. ... (closes issue #13319) Reported by: srt Patches: 13319_rtcp_manager_event_headers.diff uploaded by srt (license 378) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f4586f3018	Also make sure hinting won't crash on reload. ... (Closes issue #13312) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3a5eb27579	Remove deprecated syntax from sample config file ... (closes issue #13314) Reported by: kue git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d5c673e441	Change free to ast_free_ptr, too ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	af69ec03ed	e->data can be NULL, so use the safe version of ast_strdup() ... (closes issue #13312) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1f1dc4a0dc	Merged revisions 138027 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) | 9 lines Ensure that when a hangup occurs in autoservice, that a hangup frame gets properly deferred to be read from the channel owner when it gets taken out of autoservice. (closes issue #12874) Reported by: dimas Patches: v1-12874.patch uploaded by dimas (license 88) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	fdd92290af	Convert deprecated routines to the new names. ... (closes issue #13297) Reported by: snuffy Patches: bug13297_20080814.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	74bf61579f	That's all, folks. Not going to update the Makefile until res_jabber is ... converted (snuffy, you there? :)) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	790fde68d9	Another batch of files from RSW. The remaining apps and a few more ... files from main/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b69c8e6ab5	Another big chunk of changes from the RSW branch. Bunch of stuff from main/ ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e12e97a640	Bump a LOG_NOTICE message to LOG_DEBUG since it appears ... once for every bridged call git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	316fb598d2	Don't allow Answer() to accept a negative argument. ... Negative argument means an infinite delay and we don't want that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9b5b8246c5	Fix a calculation error I had made in the poll. The poll ... would reset to 500 ms every time a non-voice frame was received. The total time we poll should be 500 ms, so now we save the amount of time left after the poll returned and use that as our argument for the next call to poll git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ed4e6bf52b	Scrap the 500 ms delay when Asterisk auto-answers a channel. ... Instead, poll the channel until receiving a voice frame. The cap on this poll is 500 ms. The optional delay is still allowable in the Answer() application, but the delay has been moved back to its original position, after the call to the channel's answer callback. The poll for the voice frame will not happen if a delay is specified when calling Answer(). (closes issue #12708) Reported by: kactus git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	367dfcb0ab	move taskprocessor CLI commands into the core namespace ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b3970abc30	Merged revisions 136062 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame type, there are places where ast_rtp_new_source may be called where the tech_pvt of a channel may not yet have an rtp structure allocated. This caused a crash in chan_skinny, which was fixed earlier, but now the same crash has been reported against chan_h323 as well. It seems that the best solution is to modify ast_rtp_new_source to not attempt to set the marker bit if the rtp structure passed in is NULL. This change to ast_rtp_new_source also allows the removal of what is now a redundant pointer check from chan_skinny. (closes issue #13247) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	700d4501b8	Merged revisions 135949 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) | 4 lines Fix a longstanding bug in channel walking logic, and fix the explanation to make sense. (Closes issue #13124) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b2a42c3353	Merged revisions 135915 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) | 4 lines Since powerof() can return an error condition, it's foolhardy not to detect and deal with that condition. (Related to issue #13240) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	89c2844242	Merged revisions 135841,135847,135850 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@135851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	5eaf8450d6	Merged revisions 135799 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@135821 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
Kevin P. Fleming	7df8b8b848	make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3fdc96d0b4	Merged revisions 135597 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug 2008) | 1 line Use PATH_MAX for filenames ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	aca394bf0c	HTTP module memory leaks ... (closes issue #13230) Reported by: eliel Patches: res_http_post_leak.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6cf6d9eca5	Merge in changes that allow Asterisk to be built against the Hoard ... memory allocator. See doc/hoard.txt for more details. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	9051edfa4a	(closes issue #13202) ... Reported by: falves11 Tested by: murf falves11 == The changes I introduce here seem to clear up the problem for me. However, if they do not for you, please reopen this bug, and we'll keep digging. The root of this problem seems to be a subtle memory corruption introduced when creating an extension with an empty extension name. While valgrind cannot detect it outside of DEBUG_MALLOC mode, when compiled with DEBUG_MALLOC, this is certain death. The code in main/features.c is a puzzle to me. On the initial module load, the code is attempting to add the parking extension before the features.conf file has even been opened! I just wrapped the offending call with an if() that will not try to add the extension if the extension name is empty. THis seems to solve the corruption, and let the "memory show allocations" work as one would expect. But, really, adding an extension with an empty name is a seriously bad thing to allow, as it will mess up all the pattern matching algorithms, etc. So, I added a statement to the add_extension2 code to return a -1 if this is attempted. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	671627028c	Fix mime parsing by re-adding support for passing headers to callback functions ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	54b0143463	Merged revisions 134983 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul 2008) | 3 lines accomodate users who seem to lack a sense of humor :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135016 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	5aa43c0afe	Merged revisions 134883 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | 51 lines (closes issue #11849) Reported by: greyvoip Tested by: murf OK, a few days of debugging, a bunch of instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook pages of notes later, I have made the small tweek necc. to get the start time right on the second CDR when: A Calls B B answ. A hits Xfer button on sip phone, A dials C and hits the OK button, A hangs up C answers ringing phone B and C converse B and/or C hangs up But does not harm the scenario where: A Calls B B answ. B hits xfer button on sip phone, B dials C and hits the OK button, B hangs up C answers ringing phone A and C converse A and/or C hangs up The difference in start times on the second CDR is because of a Masquerade on the B channel when the xfer number is sent. It ends up replacing the CDR on the B channel with a duplicate, which ends up getting tossed out. We keep a pointer to the first CDR, and update *that* after the bridge closes. But, only if the CDR has changed. I hope this change is specific enough not to muck up any current CDR-based apps. In my defence, I assert that the previous information was wrong, and this change fixes it, and possibly other similar scenarios. I wonder if I should be doing the same thing for the channel, as I did for the peer, but I can't think of a scenario this might affect. I leave it, then, as an exersize for the users, to find the scenario where the chan's CDR changes and loses the proper start time. ........ and as to 1.4 to trunk; have I expressed my feelings about code shifting from one file to another? Good. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c95460a353	Oops, wrong define ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8b310d67b1	Merged revisions 134475 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul 2008) | 4 lines Fix a spot where a function could return without bringing a channel out of autoservice. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	853f6a8b3e	Move implementation of an attended-transfer-complete sound from one channel ... driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d86fc7fcc1	Add %u and %g to the ASTERISK_PROMPT settings, for username and group, ... respectively. Also, take the opportunity to clean up the CLI prompt generation code. (closes issue #13175) Reported by: eliel Patches: cliprompt.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	e03f7ce05a	Fix deadlock when unloading res_http_post because the uris lock was still locked. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	99db9f65b5	This commit compensates for buggy poll(2) ... implementations. Asterisk has, for a long time, had its own implementation of poll(2) which just used the input arguments to call select(2). In 1.4, this internal implementation was used for Darwin systems. This was removed in Asterisk trunk at some point, but it seems as though this was not the right move to make. On Mac OS X, it appears as though the poll used to gather CLI input does not respond properly when connecting via a remote Asterisk console. Reverting to the use of Asterisk's poll fixed the issue. Also, there is now an option for the configure script, --enable-internal-poll, which will allow for anyone to use Asterisk's internal poll implementation in case they suspect that their system's poll implementation is buggy. closes issue #11928) Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded by putnopvut (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	6291cd19bf	remove remaining Zaptel references in various places ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	06d951f585	merging the zap_and_dahdi_trunk branch up to trunk ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	6ff47b3729	actually use the cache_cache argument ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	ac79d99fa6	ast_device_state() gets called in two different ways. The first way is when ... called from elsewhere in Asterisk to find the current state of a device. In that case, we want to use the cached value if it exists. The other way is when processing a device state change. In that case, we do not want to check the cache because returning the last known state is counter productive. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c978cc1e26	Re-work comment about how device state changes are processed to be a bit more clear ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago