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Author	SHA1	Message	Date
Tilghman Lesher	ded37caf1a	Merged revisions 201531 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009) | 7 lines 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/branches/1.6.1@201532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	0a92ebc9bd	Merged revisions 201462 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines 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/branches/1.6.1@201464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	311e463f48	Merged revisions 201458 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines 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/branches/1.6.1@201460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	31ddd9da2b	Merged revisions 201453 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 | dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1644daa9aa	Merged revisions 201445 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines 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/branches/1.6.1@201448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	c33eb64920	Merged revisions 201381 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) | 16 lines 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/branches/1.6.1@201444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a3d2d156ee	Merged revisions 201344 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 | dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines 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/branches/1.6.1@201365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	1502bfdb70	Merged revisions 201262 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines 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/branches/1.6.1@201264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f5fca5c8e1	Merged revisions 201223 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	bbfd73e692	Merged revisions 201056 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines 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/branches/1.6.1@201096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	21da1c5fde	Merged revisions 201090 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines 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/branches/1.6.1@201091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	6b120bb149	Merged revisions 200985 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 | kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7 lines 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/branches/1.6.1@200989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8e5e00bd07	Merged revisions 200946 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines 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/branches/1.6.1@200987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	12e5744655	Merged revisions 200943 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) | 9 lines 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/branches/1.6.1@200947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	c4acb78d1e	Merged revisions 200764 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines 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/branches/1.6.1@200766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	a498e2f0fe	Merged revisions 200726 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines 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/branches/1.6.1@200728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	7375533824	Merged revisions 165180,200689 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines 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/branches/1.6.1@200707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	369810c36c	Merged revisions 200514 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e0aa37c39a	Merged revisions 200361 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines Merged revisions 200360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines 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/1.6.1@200363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	852a3835ee	Merged revisions 199781 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 | seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 lines Fix all of the parallel build warnings issued when running make -j#. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	713d775a87	Don't access rtp->rtcp->* if rtp->rtcp is null ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	cb76dba60a	Merged revisions 200146 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines Fix a crash due to a potentially NULL p->options. Thanks to mnicholson for pointing it out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	623b3dce51	Merged revisions 200039 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 | lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines Fix path for .flavor and .version (issue #14737) Reported by: davidw Patches: flavor.patch uploaded by davidw (license 780) Tested by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	4f775fa399	Fixes the argument order in definition of new_find_extension(). ... In the definition of new_find_extension(), the arguments 'callerid' and 'label' were swapped. The prototype declaration and all calls to the function are ordered 'callerid' then 'label', but the function itself was ordered 'label' then 'callerid'. (closes issue #15303) Reported by: JimDickenson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b95f51e4fc	Merged revisions 199958 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 | mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 lines Only try to use the invite_branch on outgoing INVITEs with auth credentials. I have added a comment to the code to help ease understanding of the logic here as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	ae7f51f26e	Merged revisions 199857 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	64af4b8465	Merged revisions 199818 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer. (closes issue #15283) Reported by: jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) Tested by: jthurman, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8e2f88caf5	Merged revisions 199743 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines module load priority This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	12cb269e94	Merged revisions 199630 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	87eda713ad	Recorded merge of revisions 199588 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, 08 Jun 2009) | 9 lines Fix a deadlock that could occur when setting rtp stats on SIP calls. (closes issue #15143) Reported by: cristiandimache Patches: 15143.patch uploaded by mmichelson (license 60) Tested by: cristiandimache ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b5e3d0c902	Merged revisions 199298 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines 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/1.6.1@199300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	64097edf92	Merged revisions 199227 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines Correct "dahdi show channels" output when specifying a group. Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	0a9c235bc1	Merged revisions 199139 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	0b9b7ffc6e	Merged revisions 199051 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines 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/1.6.1@199053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	e5be9a1b28	Blocked revisions 198958 via svnmerge ... ................ r198958 | seanbright | 2009-06-03 16:49:11 -0400 (Wed, 03 Jun 2009) | 17 lines Blocked revisions 198957 via svnmerge ........ r198957 | seanbright | 2009-06-03 16:39:10 -0400 (Wed, 03 Jun 2009) | 11 lines Fix a possible crash in pbx_spool. We were trying to reference members of a struct that had previously been freed. This patch makes sure that we free the struct after it has been removed from the spooler queue. (closes issue #15072) Reported by: garlew Patches: spool.diff uploaded by garlew (license 376) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	935853d4a3	Merged revisions 198856 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines 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 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/1.6.1@198887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a313821999	Merged revisions 198824 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines fixes issue with channels not going down after transfer Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop. (closes issue #15216) Reported by: oxymoron Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198826 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	fcdc8c20f4	Merged revisions 198791 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	62dda18b91	Merged revisions 198626 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines Add information for new meetme realtime fields ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	166dca480e	Merged revisions 198437 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) | 11 lines Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded. if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash when calling ast_unregister_timing_interface() with a NULL pointer. (closes issue #15234) Reported by: eliel Patches: timing_dahdi1.diff uploaded by eliel (license 64) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	676eaf42dc	Merged revisions 198375 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines Properly terminate the receive buffer before sending to iksemel. aji_io_recv takes the maximum number of bytes to read (instead of the total buffer size), so we have to subtract 1 from our buffer size. Without this, when we receive packets that are larger than our buffer, iksemel will choke and things get wonky. (closes issue #15232) Reported by: lp0 Patches: 05302009_res_jabber.c.patch uploaded by seanbright (license 71) Tested by: seanbright, lp0 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	3e8686a005	Merged revisions 198371 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May 2009) | 19 lines Merged revisions 198370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines Properly terminate AMI JabberSend response messages. The response message (either Error or Success) needs an extra trailing \r\n after the fields to inform the client that the message is complete. (closes issue #14876) Reported by: srt Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71) asterisk_14876.patch uploaded by srt (license 378) trunk-14876-2.diff uploaded by phsultan (license 73) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	66f4f7834d	Merged revisions 198312 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines Merged revisions 198311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines Fix a crash that occurred when MWI SMDI messages expired. (closes issue #14561) Reported by: cmoss28 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	6a3d973648	Merged revisions 198285 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 198251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we treat a missing one. (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	90dfe15ab7	Merged revisions 198248 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 | file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines When removing all packets from a dialog we also need to free the data if present. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	75b4fa611d	Merged revisions 198186 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 May 2009) | 2 lines Suggesting that only a single timing module be loaded is no longer necessary. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	08678cf81a	Merged revisions 198183 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009) | 2 lines Improve handling of trying to ACK too many timer expirations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c99a0cf74a	Merged revisions 198146 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) | 38 lines Resolve issues with choppy sound when using res_timing_pthread. The situation that caused this problem was when continuous mode was being turned on and off while a rate was set for a timing interface. A very easy way to replicate this bug was to do a Playback() from behind a Local channel. In this scenario, a rate gets set on the channel for doing file playback. At the same time, continuous mode gets turned on and off about every 20 ms as frames get queued on to the PBX side channel from the other side of the Local channel. Essentially, this module treated continuous mode and a set rate as mutually exclusive states for the timer to be in. When I dug deep enough, I observed the following pattern: 1) Set timer to tick every 20 ms. 2) Wait almost 20 ms ... 3) Continuous mode gets turned on for a queued up frame 4) Continuous mode gets turned off 5) The timer goes back to its tick per 20 ms. state but starts counting at 0 ms. 6) Goto step 2. Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick, but not most of the time. This is what produced the choppy sound (or sometimes no sound at all). Now, the module treats continuous mode and a set rate as completely independent timer modes. They can be enabled and disabled independently of each other and things work as expected. (closes issue #14412) Reported by: dome Patches: issue14412.diff.txt uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt uploaded by russell (license 2) Tested by: DennisD, russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	46d7c7d79a	Blocked revisions 198088 via svnmerge ... ........ r198088 | jpeeler | 2009-05-29 14:19:51 -0500 (Fri, 29 May 2009) | 9 lines New signaling module to handle analog operations in chan_dahdi This branch splits all the analog signaling logic out of chan_dahdi.c into sig_analog.c. Functionality in theory should not change at all. As noted in the code, there is still some unused code remaining that will be cleaned up in a later commit. Review: https://reviewboard.asterisk.org/r/253/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	c1df063c75	Merged revisions 198072 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines 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/1.6.1@198074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago