255 Commits (47ebea6a8dfa92475ad531e83b9c3d3fbad0e9c1)

Author	SHA1	Message	Date
Mark Michelson	47ebea6a8d	Fix 'd' option for app_dial and add new option to Answer application ... The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	8435535300	Tell the device state core a change happened when a channel is freed but not a specific state. ... We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	84053c05c7	Add extensions and context on manager event when new channel is created. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	3fd61d729c	Merged revisions 170648 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	dccc06063f	Merged revisions 170392 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines Fix broken call pickup There was a subtle change in ast_do_masquerade which resulted in failed attempts to pickup calls. The problem was that the value of the AST_FLAG_OUTGOING flag was copied from the clone to the original channel. In the case of call pickup, this meant that the AST_FLAG_OUTGOING flag ended up being cleared on the channel that was attempting to execute the pickup. Because this flag was not set, when ast_read came across an answer frame, it ignored it. The result of this was that the calling channel was never properly answered. This fix changes the behavior in ast_do_masquerade to set the flags on the original channel to the union of the flags on the clone channel. This way, if the AST_FLAG_OUTGOING flag is set on either of the two channels involved in the masquerade, the resulting channel will have the flag set as well. (closes issue #14206) Reported by: francesco_r Patches: 14206.patch uploaded by putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	ef6ad2b53c	Merged revisions 168561 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	859ae78977	Merged revisions 166568 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines Fix a crash resulting from a datastore with inheritance but no duplicate callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	18e07935ed	Merged revisions 166509 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9733b30ff0	Adding a new dialplan function AUDIOHOOK_INHERIT ... This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c9eb01c899	Merged revisions 164201 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines Handle a case where a call can be bridged to a channel that is still ringing. The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	035a7552d6	Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. ... (closes issue #13525) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	7fcac067b2	Merged revisions 163448 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fb242bc8fd	Fix the "failed" extension for outgoing calls. ... The conversion to use ast_check_hangup() everywhere instead of checking the softhangup flag directly introduced this problem. The issue is that ast_check_hangup() checked for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances, such as with a dummy channel. The fix is simple. Don't check tech_pvt. It's pointless, because the code path that sets this to NULL is when the channel hangup callback gets called. This happens inside of ast_hangup(), which is the same function responsible for freeing the channel. Any code calling ast_check_hangup() better not be calling it after that point, and if so, we have a bigger problem at hand. (closes issue #14035) Reported by: erogoza git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	48522988ab	In order to move away from nested function use, some changes to the recently introduced ... ast_channel_search_locked need to be made. Specifically, the caller needs to be able to pass arbitrary data which in turn is passed to the callback. This patch addresses all of the nested functions currently in asterisk trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f7c20e0dec	Merged revisions 154685 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1 line This fix was prompted by communication from user, who was seeing thousands of error logs... looks like EAGAIN. Made such uninteresting. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	086a52d9d1	Introduce a new API call ast_channel_search_locked, which iterates through the ... channel list calling a caller-defined callback. The callback returns non-zero if a match is found. This should speed up some of the code that I committed earlier today in chan_sip (which is also updated by this commit). Reviewed by russellb and kpfleming via ReviewBoard: http://reviewboard.digium.com/r/28/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	bd4eb070f3	bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 branch, and add the ones needed for all the new code here too ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	511ce6b2bf	Use the ast_str API call to reset the string instead of manually editing its internals ... (closes issue #13816) Reported by: eliel Patches: channel.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fbe13cfb86	Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() ... or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	0f0c10993c	Merged revisions 141156 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	2fceed7f6d	Merged revisions 140690 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations, where you'd want to post single-channel cdrs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	1c79a23b8e	Merged revisions 140670 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines (closes issue #13409) Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140691 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	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	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
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
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
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
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	17 years ago
Tilghman Lesher	0c23159464	Deprecate *_device_state_* APIs in favor of *_devstate_* APIs ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c780a443bf	Merged revisions 133649 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines Fix some errant device states by making the devicestate API more strict in terms of the device argument (only without the unique identifier appended). (closes issue #12771) Reported by: davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw, jvandal, murf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	d185405755	Janitor project to convert sizeof to ARRAY_LEN macro. ... (closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	bc2cfb3e81	Merged revisions 127663 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported by: murf Tested by: murf, deeperror (closes issue #12907) Reported by: falves11 Tested by: murf, falves11 (closes issue #11849) Reported by: greyvoip As to 11849, I think these changes fix the core problems brought up in that bug, but perhaps not the more global problems created by the limitations of CDR's themselves not being oriented around transfers. Reopen if necc, but bug reports are not the best medium for enhancement discussions. We need to start a second-generation CDR standardization effort to cover transfers. (closes issue #11093) Reported by: rossbeer Tested by: greyvoip, murf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	37db658b1f	Place the delay in __ast_answer prior to the channel-specific answer ... callback. This change differs from commit 127113 in that now the channel is not set to AST_STATE_UP until after the answer callback. (closes issue #12924) Reported by: snyfer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	8cdb1f7f41	change the process of inserting a delay into the ast_answer() path so that we don't tell the calling channel that it has been answered unutil after the delay; for a single-thread call this won't matter all, but for a dual-thread call (using chan_local) this may fix the problem in issue 12924 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	02b1317d0f	- add get_max_rate timing API call ... - change ast_settimeout() to honor max rate in edge cases of file playback (this will make some warning messages go away at the end of playing back a file) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	fd4a60c459	Merged revisions 125132 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it get app_rpt building again after the DAHDI changes (closes issue #12911) Reported by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	56654fc0f2	Merged revisions 123930 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008) | 5 lines Change informative messages to use the _multiple variant when multiple formats are possible. (Closes issue #12848) Reported by klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e27a98ce5a	- Fix a typo in a timing API call ... - Convert the last part of channel.c over to use the timing API. This would not have made a difference when using the dahdi timing module. I noticed it when trying to use another timing source. Oops. :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	b6457ecf4c	Merge changes from timing branch ... - Convert chan_iax2 to use the timing API - Convert usage of timing in the core to use the timing API instead of using DAHDI directly - Make a change to the timing API to add the set_rate() function - change the timing core to use a rwlock - merge a timing implementation, res_timing_dahdi Basic testing was successful using res_timing_dahdi git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	ef3b214728	Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1af7ea2df1	Merged revisions 122130 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines Occasionally, the alertpipe loses its nonblocking status, so detect and correct that situation before it causes a deadlock. (Reported and tested by ctooley via #asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	467c6f5f90	Merged revisions 121861 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines Make calls to ast_assert() actually test something, so that the error message printed is not nonsensical (reported by mvanbaak via #asterisk-bugs). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	610faf1132	Merged revisions 121442 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 lines Update BRIDGEPEER variable before we do a generic bridge in case we just broke out of a native bridge and fell through to generic. (closes issue #12815) Reported by: ramonpeek ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	85a631f767	arbitrary formatting change to test a mantis change ... (closes issue #12824) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0bee0c65b2	Minor formatting change to test a mantis change ... ... (closes issue #12824) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	90cb28f89c	Merged revisions 121280 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines Do not attempt to do emulation if an END digit is received and the length is less than the defined minimum digit length, and the other end only wants END digits (SIP INFO, for example). (closes issue #12778) Reported by: tsearle Patches: 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago