asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	eccfeaf9b4	Fix possibility of crashiness during reload in custom fields handling. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	32fdcc00b2	Standardize return values of load_config() so reload() doesn't report an error on success. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f57752d380	Leave a note about some unsafe code in cdr_manager git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	2fc4832e85	Fix version detection for API changes in spandsp. (closes issue #15355) Reported by: deuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	b390327dd7	Remove unnecessary usleep() from a couple of module unload callbacks. In passing, also tweak cdr_unregister() to hold the list lock a bit less time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	55c6789f74	Use sched_yield() instead of usleep(1) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	7339d85ee3	Blocked revisions 202022 via svnmerge ........ r202022 \| mnicholson \| 2009-06-19 16:21:15 -0500 (Fri, 19 Jun 2009) \| 4 lines Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/287/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	05da5f14d9	Merged revisions 201993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201993 \| dvossel \| 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) \| 8 lines timestamp was being converted to host order as a short rather than a long (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	25ccb5e850	Add note about the addition of calendar support git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6b53ec413d	Fix 2 typos and add support for wide character types. Reported by Benny Amorsen via the asterisk-users mailing list. http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	e85296e244	Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ad0d1bfd9e	Merged revisions 201828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201828 \| tilghman \| 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) \| 6 lines If the "h" extension fails, give it another chance in main/pbx.c. If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f341418988	One of the changes in 1.6.1 was to allow app_directory to use functionality within app_voicemail for directory functions. It is therefore no longer necessary for app_directory to be linked against the ODBC libraries (and it never was necessary for app_directory to be linked against IMAP, though it was). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e1fa477ba7	Clarify CUT code, and in the process, fix a bug in trunk only (closes issue #15320) Reported by: chappell Patches: cut_fix.patch uploaded by chappell (license 8) cut_clarify.patch uploaded by chappell (license 8) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	21ad428d0d	Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/285/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	dcfe69ec64	fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	730e60e583	Merged revisions 201600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201600 \| russell \| 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) \| 29 lines Fix memory corruption and leakage related reloads of non files mode MoH classes. For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dce6a54a4a	Trunk implementation of setting an alternate RTP source. This contains the interface by which we can let an rtp instance know that it might start receiving audio from a new source. This is similar in nature to revision 197588 of Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a11ac5ae2f	parsing extension correctly from sip register lines If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'. (closes issue #15111) Reported by: ffs Patches: chan_sip.c_register-parser.patch uploaded by ffs (license 730) Tested by: ffs, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	68ba81dfe6	Add rtsavesysname to chan_iax chan_sip has an option to save the sysname on rtupdate. This patch copies that same logic to chan_iax. (closes issue #14837) Reported by: barthpbx Patches: iax2-rtsavesysname.patch uploaded by barthpbx (license 744) rt_iax.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	7b37a60418	Initialize additional variables, to prevent a possible crash. (closes issue #15186) Reported by: ajohnson Patches: 20090528__issue15186.diff.txt uploaded by tilghman (license 14) Tested by: ajohnson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	99b98d8f9a	Fix problem with no audio due to ignoring the SDP. A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d8cc968adc	Merged revisions 201450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 \| mmichelson \| 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) \| 9 lines 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/trunk@201458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	82e4b25230	ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	de8fae17d4	Merged revisions 201423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201423 \| dvossel \| 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) \| 19 lines StopMixMonitor race condition (not giving up file immediately) StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	ecfbab0782	Merged revisions 201380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 \| dbrooks \| 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) \| 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	9bf67151c9	SIP registry ref count error During a sip reload, the list of sip_registry objects are supposed to be traversed, unlinked, and destroyed, but destruction never takes place due to a ref counting error. This causes a memory leak when registry items are removed from sip.conf and reloaded. While the registries are removed from the global list, they are not removed from the scheduler. Because of this, SIP register attempts continue to be sent out for the item even though it may no longer be in the .conf. (closes issue #15295) Reported by: amorsen Review: https://reviewboard.asterisk.org/r/282/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	940accbd99	update chan_iax to use 64bit feature flags. (closes issue #15335) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/284/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	962bd7ab26	Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 \| kpfleming \| 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) \| 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is not, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	9a66b1dcdf	fix issue with build_contact introduced by the "SIP trasnport type issues" commit git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a342fb0700	Update my e-mail address (thanks for the props, russell :)) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f1dc620467	Enable applications to enable/disable digit and tone detection. Some applications (notably app_fax) do not need digit detection nor FAX tone detection while they are running, and if Asterisk is using software DSPs to provide the detection, this consumes extra CPU cycles that could be better spent on the actual application. This patch allows applications to query and control the state of digit and tone detection on a channel, and modifies app_fax to disable them while the FAX operations are occurring (and re-enable digit detection afterwards). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	5b2939916f	Explicitly test for 'static weakref' support. Since we use 'static' weakref symbols, and not all GCC versions support them, test for that combination explicitly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	ab73c3aabe	When compiling in an Emacs-spawned shell, always print directory names. This change ensures that Emacs can find the proper source files when parsing compiler error messages, since it uses the 'make' output including directory names to do it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4a93d3bcb5	Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4c0265664e	Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 \| kpfleming \| 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) \| 11 lines 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/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	10ea4910e7	Fix problems with new compiler attribute checking in configure script. The last changes to ast_gcc_attribute.m4 caused some problems checking for various attributes, because the scope of the symbol the attribute is applied to can be important; this patch allows the scope to be specified for the check. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ee8cdd555f	SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	4271603247	add FILE_STORAGE to Voicemail Build Options Voicemail can only use one storage module at the moment. Because it's unclear that selecting one of the storage modules in menuselect will disable filesystem storage we now have a FILE_STORAGE option that conflicts with the other modules. (closes issue #15333) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fc6ec8a16a	Add Sean Bright to CREDITS - Thanks, Sean! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	6031389b43	Recorded merge of revisions 200875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200875 \| eliel \| 2009-06-16 09:25:51 -0400 (Tue, 16 Jun 2009) \| 5 lines Show the interface name on error, if it is not found. If the smdiport specified is not found, show the interface name instead of '(null)'. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	a179e144cd	Show the interface name on error, if it is not found. If the smdiport specified is not found, show the interface name instead of '(null)'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	3c23511069	Don't claim a char * is a mansession object. Since there was only 1 bucket, and no hash function was specified, the code actually worked perfectly fine. However, in theory, this was invalid use of the OBJ_POINTER flag, so remove it so the code provides a better usage example. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	2c8cd1db92	keep backwards compatible chan_dahdi with older openr2 versions by not using the new skip category feature unless supported git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4af5dd813c	Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	dcc651d99b	Add missing closure of verbatim environment. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	5b79acdb42	Document the new automatic 'ignoresdpversion' behavior. Asterisk will now automatically ignore incorrect incoming SDP version numbers when necessary to complete a T.38 re-INVITE operation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	85e57521ab	Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is not set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	aaeec3b40f	Last batch of 'static' qualifiers for module-level global variables. Fix up modules in the 'apps' directory, and also correct the bad example of enum definitions in include/asterisk/app.h, which many developers followed (thanks for reading the documentation!). In addition, add some basic usage examples of the 'pahole' and 'pglobal' tools to the coding guidelines. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	82fb56886e	More 'static' qualifiers on module global variables. The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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