asterisk

Commit Graph

Author	SHA1	Message	Date
Michiel van Baak	8edfe07e6d	make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	05899c19a1	Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links before any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	d5cc6f0879	Make sure 'start' is always initialized. This is the same as rev 216222 in trunk but 1.4 is affected as well git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ed1951d895	Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	fafa668b04	Modify comment to be a bit more accurate. We have kept this comment around long enough, that it's pretty clear that we're keeping the code, because changing the code would require a pretty fundamental architectural shift. We've also taken criticism in some quarters, because it was believed that it was referring to the code being nasty. No, the code isn't nasty, just the operation itself is rather odd. Fixed for eternity (probably not). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	899272c2cc	ast_write() ignores ast_audiohook_write() results In ast_write(), if a channel has a list of audiohooks, those lists are written to and the resulting frame is what ast_write() should continue with. The problem was the returned audiohook frame was not being handled at all, and the original frame passed into it did not contain the mixed audio, so essentially audio was being lost. One result of this was chan_spy's whisper mode no longer worked. To complicate the issue, frames passed into ast_write may either be a single frame, or a list of frames. So, as the list of frames is processed in the audiohook_write, the returned frames had to be added to a new list. (closes issue #15660) Reported by: corruptor Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a45f5e5be0	I should always compile before committing... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	10a972f98e	Fix pronunciation of German dates. (closes issue #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded by Benjamin Kluck (license 803) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0055593341	Improve error message by informing user exactly which function is missing a parethesis. (closes issue #15242) Reported by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by loloski (license 68) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a4d1e6120f	Delay the creation of temporary files until we have a valid manager command to handle. Without this patch, asterisk creates a temporary file before determining if the specified command is valid. If invalid, we weren't properly cleaning up the file. (closes issue #15730) Reported by: zmehmood Patches: M15730.diff uploaded by junky (license 177) Tested by: zmehmood git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@212763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	63cc189747	AST-2009-005 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	541976843e	Small oops. Clear the flags which have been checked. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e87d76cb94	Because channel information can be accessed outside of the channel thread, we must lock the channel prior to modifying it. (closes issue #15397) Reported by: caspy Patches: 20090714__issue15397.diff.txt uploaded by tilghman (license 14) Tested by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ca0f026f41	Reverting index() fix, applying a different methodology, based upon developer discussions. (related to issue #15639) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f5a5763ee9	Helps if we export the index() function. (Related to issue #15639) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a70128e190	Apparently, some platforms don't have the index() function. (closes issue #15639) Reported by: nmav git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ffe395f410	Resolve a valgrind warning about a read from uninitialized memory. (issue #15396) Reported by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b5bea3704c	Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	361c9a99e1	Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	f622e06bbe	Fix logic errors from 208746 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	fc5db2b241	Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	98dcd8946e	Export symbols for functions included in our compatibility headers. (closes issue #15556) Reported by: smw1218 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e0827ae778	Document default timeout for AMI originations. AST-224 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	75f1eaf2a1	Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker after all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided either in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8b67a33369	Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	b1fe655954	avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	df4d75f4a7	Print CID match in "show dialplan". (closes issue #14702) Reported by: klaus3000 Patches: patch_asterisk_1.4.23_CID_matching.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	beaf6217b3	Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	2e330f772c	moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0b1f3adf7f	Add redirection warnings for the invalid language codes previously removed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	bdada0dce1	moving device state functions from pbx.h to devicestate.h to sync with other branches git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	4c99b19973	Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e8f0570118	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/branches/1.4@204556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	b3e413e910	Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	60270012a9	"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/branches/1.4@204469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fe7923abfc	Don't fast forward past the end of a message. This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	5e3e234df6	I didn't see that Mark already fixed the underlying issue! Yay for removing useless code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	03d3fb7a7a	Don't try to free NULL git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	140d84dfc5	Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	72b285ed96	Report CallerID change during a masquerade. Reported by: markster git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	03909de702	Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	94fa4d11b5	Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	af2e7f5eab	Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d3bea6da02	Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	7605487610	Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ddb4e3f2e7	Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and res_feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	aa2fd9a4c2	Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	590408dca3	Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3268149a1f	Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	829907e467	Fix broken attended transfers The bridge was terminating immediately after the attended transfer was completed. The problem was because upon reentering ast_channel_bridge nexteventts was checked to see if it was set and if so could possibly return AST_BRIDGE_COMPLETE. (closes issue #15183) Reported by: andrebarbosa Tested by: andrebarbosa, tootai, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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1107 Commits (3913dcbc40039ca3c1942b7300fc45daeb4970ea)