156 Commits (19afbc2af2a59cb07915e52db9dce5ccb916a6f7)

Author	SHA1	Message	Date
Automerge script	8aef9fdedd	Merged revisions 369005 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369005 | kpfleming | 2012-06-15 11:07:08 -0500 (Fri, 15 Jun 2012) | 22 lines Multiple revisions 369001-369002 ........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	f3f2ac35f4	Multiple revisions 365155,365160,365299,365320,365399,365475,365478,365575,365632,365701,365898,365990,366049,366053,366106,366168,366241,366297,366390,366412,366591,366598,366741,366792,366881,366884,366948,367003,367028,367267,367299,367369,367417,367470,367562,367679,367731,367782,367844,367907,367978,367981,368042,368093,368267,368310,368407,368470,368499,368524,368536,368568,368587 ... ........ r365155 | may | 2012-05-03 09:27:00 -0500 (Thu, 03 May 2012) | 11 lines Fix coverity static analysis warning, allocate full ie structure instead of without data buffer (close issue ASTERISK-19674) Reported by: Matt Jordan Patches: ASTERISK-19674.patch (License #5415) ........ Merged revisions 365143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365160 | may | 2012-05-03 10:01:14 -0500 (Thu, 03 May 2012) | 11 lines Fix warning of Coverity Static analysis, change H225ProtocolIdentifier from value to pointer per functions that use this. (close issue ASTERISK-19670) Reported by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415) ........ Merged revisions 365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365299 | mmichelson | 2012-05-04 10:51:04 -0500 (Fri, 04 May 2012) | 12 lines Fix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report. These three all are in RTP code that attempts to print the number of sequence number cycles in an RTCP RR report. The code was masking out the upper 16 bits and then shifting the number right by 16 bits. This led to an all zero result in all cases. The fix is to do the shift without the bit masking. (issue ASTERISK-19649) ........ Merged revisions 365298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365320 | rmudgett | 2012-05-04 11:28:06 -0500 (Fri, 04 May 2012) | 30 lines Fix local channel chains optimizing themselves out of a call. * Made chan_local.c:check_bridge() check the return value of ast_channel_masquerade(). In long chains of local channels, the masquerade occasionally fails to get setup because there is another masquerade already setup on an adjacent local channel in the chain. * Made the outgoing local channel (the ;2 channel) flush one voice or video frame per optimization attempt. * Made sure that the outgoing local channel also does not have any frames in its queue before the masquerade. * Made do the masquerade immediately to minimize the chance that the outgoing channel queue does not get any new frames added and thus unconditionally flushed. * Made block indication -1 (Stop tones) event when the local channel is going to optimize itself out. When the call is answered, a chain of local channels pass down a -1 indication for each bridge. This blizzard of -1 events really slows down the optimization process. (closes issue ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec Davis Review: https://reviewboard.asterisk.org/r/1894/ ........ Merged revisions 365313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365399 | kmoore | 2012-05-04 17:15:05 -0500 (Fri, 04 May 2012) | 13 lines Fix many issues from the NULL_RETURNS Coverity report Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365475 | mjordan | 2012-05-07 13:39:10 -0500 (Mon, 07 May 2012) | 20 lines Support VoiceMail d() option when extension does not exist in channel's context The VoiceMail d([c]) option is documented to accept digits for a new extension in context <c>, if played during the greeting. This option works fine if the extension being redirected to has an extension with the same initial digit in the channel's current context. If that digit did not happen to exist in some extension, a dialplan match would fail and the user would not be redirected. This patch fixes it such that if the <c> option is used, the extensions are matched in that context as opposed to the caller's original context. (closes issue ASTERISK-18243) Reported by: mjordan Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1892 ........ Merged revisions 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365478 | rmudgett | 2012-05-07 13:43:08 -0500 (Mon, 07 May 2012) | 5 lines Fix type punned compiler warning in test_config.c ........ Merged revisions 365476 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365575 | mmichelson | 2012-05-08 10:51:13 -0500 (Tue, 08 May 2012) | 22 lines Send more accurate identification information in dialog-info SIP NOTIFYs. This uses the calling channel's caller ID and connected line information to populate the remote and local identities in the dialog-info NOTIFY when an extension is ringing. There is a bit of an oddity here, and that is that we seed the remote target with the To header of the outbound call rather than the from header. This is because it was reported that seeding with the from header caused hints to be broken with certain SNOM devices. A comment has been added to the code to explain this. (closes issue ASTERISK-16735) reported by Maciej Krajewski patches: local_remote_hint2.diff uploaded by Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark Michelson (license #5049) Tested by Niccolo Belli ........ Merged revisions 365574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365632 | rmudgett | 2012-05-08 13:08:01 -0500 (Tue, 08 May 2012) | 13 lines * Fix accept/decline DTMF buffer overwrite in FollowMe. * Made use MAX_YN_STRING define to make all accept/decline DTMF buffers the same size. Just using 20 isn't good enough when someone didn't get the memo. * Fix stupid use of a global variable in FollowMe. (ynlongest) * Fix bit field declarations in FollowMe. ........ Merged revisions 365631 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365701 | rmudgett | 2012-05-08 15:25:08 -0500 (Tue, 08 May 2012) | 12 lines * Fix FollowMe memory leak on error paths in app_exec(). * Fix FollowMe leaving recorded caller name file on error paths in app_exec(). * Use correct buffer dimension define in struct call_followme.moh[] and struct fm_args.namerecloc[]. This fixes unexpected namerecloc filename length restriction. ........ Merged revisions 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365898 | mmichelson | 2012-05-09 11:15:28 -0500 (Wed, 09 May 2012) | 29 lines Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. chan_sip was coded under the assumption that a SIP dialog with an owner channel will always be destroyed after the owner channel has been hung up. However, there are situations where the SIP dialog can time out and auto destruct before the corresponding channel has hung up. A typical example of this would be if the 'h' extension in the dialplan takes a long time to complete. In such cases, __sip_autodestruct() would complain about the dialog being auto destroyed with an owner channel still in place. The problem is that even once the owner channel was hung up, the sip_pvt would still be linked in its ao2_container because nothing would ever unlink it. The fix for this is that if __sip_autodestruct() is called for a sip_pvt that still has an owner channel in place, the destruction is rescheduled for 10 seconds in the future. This will continue until the owner channel is finally hung up. (closes issue ASTERISK-19425) reported by David Cunningham Patches: ASTERISK-19425.patch uploaded by Mark Michelson (License #5049) (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by Dean Vesvuio ........ Merged revisions 365896 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r365990 | jrose | 2012-05-09 14:12:32 -0500 (Wed, 09 May 2012) | 18 lines Block on frameout if the hardware has enough samples to complete a frame. Fixes some problems with skipping audio in elaborate scenarios involving multiple codecs by making codec_dahdi operate in a more synchronous fashion similar to codec_g729. This change also fixes the use of file conversion tools from Asterisk's CLI. This change may cause the thread responsible for transcoding audio to block briefly (Shaun Ruffell describes this as 'several milliseconds') while waiting for the hardware transcoder. (closes issue ASTERISK-19643) reported by: Shaun Ruffell Patches: 0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch uploaded by Shaun Ruffell (license 5417) ........ Merged revisions 365989 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366049 | jrose | 2012-05-10 10:43:06 -0500 (Thu, 10 May 2012) | 9 lines Coverity Report: Fix issues for error type UNINIT in Core supported modules (issue ASTERISK-19652) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/ ........ Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366053 | mmichelson | 2012-05-10 11:13:06 -0500 (Thu, 10 May 2012) | 9 lines Close the proper tcptls_session when session creation fails. (issue AST-998) Reported by: Thomas Arimont Tested by: Thomas Arimont ........ Merged revisions 366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366106 | jrose | 2012-05-10 11:55:22 -0500 (Thu, 10 May 2012) | 9 lines Coverity Report: Fix issues for error type CHECKED_RETURN for core (issue ASTERISK-19658) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1905/ ........ Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366168 | kmoore | 2012-05-10 15:54:08 -0500 (Thu, 10 May 2012) | 13 lines Resolve FORWARD_NULL static analysis warnings This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366241 | rmudgett | 2012-05-10 18:42:43 -0500 (Thu, 10 May 2012) | 7 lines * Made ast_change_name() hold the channels container lock while changing the channel name. * Eliminate redundant list not empty check in clone_variables(). ........ Merged revisions 366240 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366297 | russell | 2012-05-11 18:59:35 -0500 (Fri, 11 May 2012) | 19 lines format_mp3: Fix a possible crash mp3_read(). This patch fixes a potential crash in mp3_read() by not assuming that dbuf has enough data to finish filling up the output buffer. The patch also makes sure that the dbuf state gets reset after we know we read everything out of it already. In passing, this patch includes some other cleanups of this module, including stripping trailing whitespace, formatting fixes based on coding guidelines, and removing a number of unused members from the private state struct. (closes issue ASTERISK-19761) Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk ........ Merged revisions 366296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366390 | mmichelson | 2012-05-14 14:16:36 -0500 (Mon, 14 May 2012) | 25 lines Fix broken reinvite glare scenario. To make a long story short, reinvite glares were broken because Asterisk would invert the To and From headers when ACKing a 491 response. The reason was because the initreq of the dialog was being changed to the incoming glared reinvite instead of being set to the outgoing glared reinvite. This change has three parts * In handle_incoming, we never will reject an ACK because it has a to-tag present, even if we think the request may be out of dialog. * In handle_request_invite, we do not change the initreq when receiving a reinvite to which we will respond with a 491. * In handle_request_invite, several superflous settings up pendinginvite have been removed since this is dones automatically by transmit_response_reliable Review: https://reviewboard.asterisk.org/r/1911 ........ Merged revisions 366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366412 | mmichelson | 2012-05-14 15:06:58 -0500 (Mon, 14 May 2012) | 19 lines Fix two more coverity constant expression result findings. These correspond to findings 0 and 1 in the core findings of ASTERISK-19649. After contacting Mark Spencer, he was unsure of what the intent behind these lines of code were, so they are being axed. For Asterisk 1.8 and 10, the output of debugging DUNDi frames will not be changed, but for trunk the "Retry" portion will be omitted since it does not properly distinguish retransmissions from initial frames. (closes issue ASTERISK-19649) Reported by Matthew Jordan ........ Merged revisions 366409 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366591 | jrose | 2012-05-15 15:44:59 -0500 (Tue, 15 May 2012) | 15 lines chan_sip: Check the right channel's host address for directmediapermit/deny Prior to this patch, when checking the addresses for directmediapermit and denydirectmediadeny, Asterisk would check the host address of the channel permit/deny was specified, which defers from the expectations of both our users and the development team. Instead, directmediapermit/deny now checks against the address of the channel that the peer with the ACL is connected to. (issue AST-876) Review: https://reviewboard.asterisk.org/r/1899/ ........ Merged revisions 366547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366598 | mmichelson | 2012-05-15 18:39:06 -0500 (Tue, 15 May 2012) | 8 lines Correct misuse of ast_strip_quoted() when getting a Diversion header's reason parameter. The use here was assuming that the pointer would be updated, but the updated string is actually returned by ast_strip_quoted() instead. ........ Merged revisions 366597 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366741 | mjordan | 2012-05-17 07:57:30 -0500 (Thu, 17 May 2012) | 23 lines Fix checking bounds of array index after using it; improper sizeof This patch fixes two problems pointed out by a static analysis tool. * In chan_dahdi, when an event is handled the index of the sub channel is first obtained. In very off nominal cases, the method that determines the index can return a negative value. In the event handling code, whether or not the index returned is valid was being checked after that value was used to index into an array. This patch makes it so the value is checked before any indexing is done. * In res_calendar_ews, sizeof was being passed a pointer instead of the struct to determine the amount of memory to allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes issue ASTERISK-19671) Reported by: Matt Jordan ........ Merged revisions 366740 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366792 | jrose | 2012-05-17 09:41:13 -0500 (Thu, 17 May 2012) | 10 lines chan_sip: Fix missed locking of opposing pvt for directmedia acl from r366547 It also required deadlock avoidance since two sip_pvts structs needed to be locked simultaneously. Trunk handles it differently, so this is a 1.8 and 10 patch only. ........ (issue AST-876) Merged revisions 366791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366881 | mjordan | 2012-05-18 09:01:56 -0500 (Fri, 18 May 2012) | 65 lines Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366884 | kmoore | 2012-05-18 09:18:47 -0500 (Fri, 18 May 2012) | 9 lines Reorder and renumber tests appropriately It appears that a patch did not apply properly when adding tests 12 and 13 and test 11 was duplicated. These tests have been reordered and renumbered such that they make sense. ........ Merged revisions 366882 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r366948 | mjordan | 2012-05-18 10:45:42 -0500 (Fri, 18 May 2012) | 20 lines Fix more memory leaks This patch adds to what was fixed in r366880. Specifically, it addresses the following: * chan_sip: dispose of an allocated frame in off nominal code paths in sip_rtp_read * func_odbc: when disposing of an allocated resultset, ensure that any rows that were appended to that resultset are also disposed of * cli: free the created return string buffer in another off nominal code path (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922/ ........ Merged revisions 366944 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367003 | mmichelson | 2012-05-18 12:00:14 -0500 (Fri, 18 May 2012) | 19 lines Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX structures were allocated but never freed. This was a bigger issue for clients than servers since new SSL_CTX structures could be allocated for each connection. Servers, on the other hand, typically set up a single SSL_CTX for their lifetime. This is solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is freed so that a new one can take its place. 2. A companion to ast_ssl_setup() called ast_ssl_teardown() has been added so that servers can properly free their SSL_CTXs. (issue ASTERISK-19278) ........ Merged revisions 367002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367028 | mmichelson | 2012-05-18 12:50:18 -0500 (Fri, 18 May 2012) | 18 lines Address MISSING_BREAK static analysis reports some more. This addresses core findings 4 and 6. Moises Silva helped me by stating that a break could be safely added to the case where it is added in chan_dahdi.c In say.c, I have added a comment indicating that static analysis complains but that it is currently unknown if this is correct. This fixes all core findings of this type. (closes issue ASTERISK-19662) reported by Matthew Jordan ........ Merged revisions 367027 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367267 | twilson | 2012-05-22 11:17:46 -0500 (Tue, 22 May 2012) | 14 lines Resolve crash in subscribing for MWI notifications ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable should definitely not be used after that. To solve this in the two cases that affect subscribing for MWI notifications, we instead save the ref locally, and unref them in the error conditions. (closes issue ASTERISK-19827) Reported by: B. R Review: https://reviewboard.asterisk.org/r/1940/ ........ Merged revisions 367266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367299 | twilson | 2012-05-22 12:21:51 -0500 (Tue, 22 May 2012) | 21 lines Fix race condition for CEL LINKEDID_END event This patch fixes to situations that could cause the CEL LINKEDID_END event to be missed. 1) During a core stop gracefully, modules are unloaded when ast_active_channels == 0. The LINKDEDID_END event fires during the channel destructor. This means that occasionally, the cel_* module will be unloaded before the channel is destroyed. It seemed generally useful to wait until the refcount of all channels == 0 before unloading, so I added a channel counter and used it in the shutdown code. 2) During a masquerade, ast_channel_change_linkedid is called. It calls ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids container in cel.c. It didn't ref the new linkedid. Now it does. Review: https://reviewboard.asterisk.org/r/1900/ ........ Merged revisions 367292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367369 | mjordan | 2012-05-23 08:25:04 -0500 (Wed, 23 May 2012) | 26 lines Re-add LastMsgsSent value for SIP peers Previously, MWI logic utilized a counter called 'lastmsgssent' to know whether or not MWI NOTIFY requests had been sent to a specific peer. When MWI notifications were changed to use the internal event framework, this value was no longer needed for its original purpose. Hence, it was no longer updated with the new/old message counts for a peer. The value was previously removed for Asterisk 10; however, since it was still present in Asterisk 1.8 and still useful for reporting purposes, it was decided to re-add the value. This patch re-adds the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show peer [peer]' command, and makes it so that the value of lastmsgssent is updated appropriately. The value should now display the new/old message counts for a particular peer. (closes issue ASTERISK-17866) Reported by: Steve Davies patches by: ast-17866-rb1272.patch (License #5041 by irroot) Modified slightly for this commit Review: https://reviewboard.asterisk.org/r/1939 ........ Merged revisions 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367417 | mmichelson | 2012-05-23 15:29:03 -0500 (Wed, 23 May 2012) | 7 lines Only call SSL_CTX_free if DO_SSL is defined. Thanks to Paul Belanger for pointing out this error. ........ Merged revisions 367416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367470 | rmudgett | 2012-05-23 18:16:49 -0500 (Wed, 23 May 2012) | 9 lines Fix WaitExten(x,m(musicclass)) string termination. The AST_CONTROL_HOLD MOH class from the WaitExten application can now be queued onto a channel, passed over local channels with the /m option, and passed over IAX channels. ........ Merged revisions 367469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367562 | mjordan | 2012-05-24 08:32:33 -0500 (Thu, 24 May 2012) | 24 lines Fix crash in ConfBridge when user announcement is played for more than 2 users A patch introduced in r354938 made it so that ConfBridge would not attempt to play sound files if those files did not exist. Unfortunately, ConfBridge uses the same underlying function, play_sound_helper, to playback both sound files and numbers to callers. When a number is being played back, the name of the sound file is expected to be NULL. This NULL value was passed into a function that tested for the existance of a sound file and is not tolerant to NULL file names, causing a crash. This patch fixes the behavior, such that if a sound file does not exist we do not attempt to play it, but we only attempt that check if the a sound file was specified in the first place. If a sound file was not specified, we use the 'play number' logic in the helper function. (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested by: Florian Gilcher patches: asterisk-19899.diff uploaded by mjordan (license 6283) ........ r367679 | rmudgett | 2012-05-24 17:29:23 -0500 (Thu, 24 May 2012) | 34 lines Fix Dial I option ignored if dial forked and one fork redirects. The Dial and Queue I option is intended to block connected line updates and redirecting updates. However, it is a feature that when a call is locally redirected, the I option is disabled if the redirected call runs as a local channel so the administrator can have an opportunity to setup new connected line information. Unfortunately, the Dial and Queue I option is disabled for *all* forked calls if one of those calls is redirected. * Make the Dial and Queue I option apply to each outgoing call leg independently. Now if one outgoing call leg is locally redirected, the other outgoing calls are not affected. * Made Dial not pass any redirecting updates when forking calls. Redirecting updates do not make sense for this scenario. * Made Queue not pass any redirecting updates when using the ringall strategy. Redirecting updates do not make sense for this scenario. * Fixed deadlock potential with chan_local when Dial and Queue send redirecting updates for a local redirect. * Converted the Queue stillgoing flag to a boolean bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1920/ ........ Merged revisions 367678 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367731 | elguero | 2012-05-24 21:29:26 -0500 (Thu, 24 May 2012) | 20 lines Fix pvt_sip for inbound call to use peer's allowtransfer setting The pvt_sip allowtransfer was not being set to that of the peer's setting. Therefore, the global allowtransfer setting was being used instead which would lead to calls not being transfered if the global setting was set to 'no' despite the setting on the peer being 'yes' and vice versa, calls would be allowed to transfer even if the peer's setting was 'no' but the global setting was 'yes'. (Closes issue ASTERISK-19856) Reported by: Jacek Tested by: Michael L. Young, Jacek Patches: issue-asterisk-19856-branch10-v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1923/ ........ Merged revisions 367730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367782 | rmudgett | 2012-05-25 11:30:55 -0500 (Fri, 25 May 2012) | 18 lines AST-2012-007: Fix IAX receiving HOLD without suggested MOH class crash. * Made schedule_delivery() set the received frame f->data.ptr to NULL if the datalen is zero. * Fix queue_signalling() memcpy() size error. * Made queue_signalling() not use C++ keyword variable names. (closes issue ASTERISK-19597) Reported by: mgrobecker Patches: jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Michael L. Young ........ Merged revisions 367781 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367844 | mjordan | 2012-05-29 13:33:20 -0500 (Tue, 29 May 2012) | 21 lines AST-2012-008: Fix remote crash vulnerability in chan_skinny When a skinny session is unregistered, the corresponding device pointer is set to NULL in the channel private data. If the client was not in the on-hook state at the time the connection was closed, the device pointer can later be dereferened if a message or channel event attempts to use a line's pointer to said device. The patches prevent this from occurring by checking the line's pointer in message handlers and channel callbacks that can fire after an unregistration attempt. (closes issue ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: mjordan, Damien Wedhorn Patches: AST-2012-008-1.8.diff uploaded by mjordan (license 6283) AST-2012-008-10.diff uploaded by mjordan (licesen 6283) ........ r367907 | rmudgett | 2012-05-29 17:28:55 -0500 (Tue, 29 May 2012) | 17 lines Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules) * Fix only issue pointed out by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). * Change use of %i to %d in sscanf() in find_user(). The use of %i gives unexpected parsing because it can accept hex, octal, and decimal integer formats. * Changed other uses of %i in app_meetme() to use %d for consistency. (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 367906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367978 | rmudgett | 2012-05-30 12:39:24 -0500 (Wed, 30 May 2012) | 19 lines Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" commands. * Fix sig_pri_lock_owner() to avoid deadlock properly. * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid deadlock properly. * Code ss7_grab() better. (closes issue ASTERISK-19854) Reported by: Jaxon Patches: jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7) Tested by: Jaxon ........ Merged revisions 367976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r367981 | rmudgett | 2012-05-30 13:07:28 -0500 (Wed, 30 May 2012) | 7 lines Use the DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854) ........ Merged revisions 367980 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368042 | rmudgett | 2012-05-31 13:20:15 -0500 (Thu, 31 May 2012) | 10 lines Coverity Report: Fix issues for error type REVERSE_INULL (core modules) * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 368039 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368093 | elguero | 2012-05-31 22:28:09 -0500 (Thu, 31 May 2012) | 17 lines Add documentation to function CHANNEL for options echocan_mode and buffers The ability to set "echocan_mode" and "buffers" through the dialplan was added to chan_dahdi some time ago. This patch adds some documentation to func_channel. (Closes issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael L. Young Patches: asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ ........ Merged revisions 368092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368267 | kpfleming | 2012-06-01 15:22:44 -0500 (Fri, 01 Jun 2012) | 20 lines Improve SDP parsing warning messages * 'Unsupported media type' is only reported when that is in fact the case, not when a supported media type is included in an 'm' line that has an invalid format. * All warning messages related to parsing 'm' lines now include the 'm' line contents. * (minor bugfix) newline added to port-number-zero warning messages. * Warning messages improved to use RFC-specified terminology for various items. * Warnings for offers that include more than one port for a single media type now include the media type. Review: https://reviewboard.asterisk.org/r/1811/ ........ Merged revisions 368218 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368310 | rmudgett | 2012-06-01 18:24:25 -0500 (Fri, 01 Jun 2012) | 15 lines Fix deadlock when Gosub used with alternate dialplan switches. Attempting to remove a channel from autoservice with the channel lock held will result in deadlock. * Restructured gosub_exec() to not call ast_parseable_goto() and ast_exists_extension() with the channel lock held. (closes issue ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368407 | rmudgett | 2012-06-04 14:08:52 -0500 (Mon, 04 Jun 2012) | 23 lines Fix potential deadlock between masquerade and chan_local. * Restructure ast_do_masquerade() to not hold channel locks while it calls ast_indicate(). * Simplify many calls to ast_do_masquerade() since it will never return a failure now. If it does fail internally because a channel driver callback operation failed, the only thing ast_do_masquerade() can do is generate a warning message about strange things may happen and press on. * Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This change fixes half of the deadlock reported in ASTERISK-19801 between masquerades and chan_iax. (closes issue ASTERISK-19537) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1915/ ........ Merged revisions 368405 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368470 | rmudgett | 2012-06-04 16:11:42 -0500 (Mon, 04 Jun 2012) | 10 lines Document BLINDTRANSFER behavior change. (issue ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call ........ Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368499 | mmichelson | 2012-06-04 17:02:26 -0500 (Mon, 04 Jun 2012) | 16 lines Relay proper SIP responses on calling side. Revision 351130 broke corect HANGUPCAUSE setting for the 404 case in chan_sip. Other cases were also potentially broken. This patch fixes the relaying of causes to be what they used to be. (closes issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter Doekes (via a reviewboard test to be committed later) Patches: chan_sip.diff uploaded by Pavel Troller (license #6302) ........ Merged revisions 368498 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368524 | kmoore | 2012-06-05 10:19:58 -0500 (Tue, 05 Jun 2012) | 11 lines Ensure that pages and emails are sent using RFC822-compliant date format When localization was added to app_voicemail, these headers were altered when they should have remained in en_US format for RFC compliance. This reverts the changes to those two lines. (closes issue ASTERISK-19876) ........ Merged revisions 368520 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368536 | kmoore | 2012-06-05 10:27:01 -0500 (Tue, 05 Jun 2012) | 8 lines Resolve some build warnings My newly upgraded compiler caught these usages of uninitialized values. They weren't actually used. ........ Merged revisions 368533 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368568 | rmudgett | 2012-06-05 20:10:10 -0500 (Tue, 05 Jun 2012) | 15 lines Fix parked call performing a DTMF blind transfer after being retrieved. When a parked call was retrieved from the parking lot, it could not do a blind transfer because it caused the involved calls to be hung up unconditionally. * Made the ParkedCall application return the ast_bridge_call() return value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc ........ Merged revisions 368567 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368587 | kmoore | 2012-06-06 11:09:10 -0500 (Wed, 06 Jun 2012) | 12 lines Ensure overlapping hold flags do not conflict When changing between different modes of hold, the flags were not being cleared out properly causing a failure to change hold states. (closes issue ASTERISK-19919) Patch-by: Morten Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365155,365160,365299,365320,365399,365475,365478,365575,365632,365701,365898,365990,366049,366053,366106,366168,366241,366297,366390,366412,366591,366598,366741,366792,366881,366884,366948,367003,367028,367267,367299,367369,367417,367470,367562,367679,367731,367782,367844,367907,367978,367981,368042,368093,368267,368310,368407,368470,368499,368524,368536,368568,368587 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368781 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	978a506d68	Make CLI "core show channel" not hold the channel lock during console output. ... Holding the channel lock while the CLI "core show channel" command is executing can slow down the system. It could block the system if the console output is halted or paused. * Made capture the CLI "core show channel" output into a buffer to be output after the channel is unlocked. * Removed use of C++ keyword as a variable name. out renamed to obuf. * Checked allocation of obuf for failure so will not crash. (closes issue ASTERISK-18571) Reported by: Pavel Troller Tested by: rmudgett ........ Merged revisions 344661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	05ddffccc4	Merged revisions 313860 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313860 | jrose | 2011-04-15 10:08:05 -0500 (Fri, 15 Apr 2011) | 17 lines Merged revisions 313859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) | 10 lines Fix a Tab Completion bug that occurs due to multiple matches on a substring. Makes word_match function in cli.c repeat a search for a command string until a proper match is found or the string is searched to the last point. (closes issue #17494) Reported by: ffossard Review: https://reviewboard.asterisk.org/r/1180/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	530afe7d97	Merged revisions 313366 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) | 2 lines Added "Connected Line ID" and "Connected Line ID Name" to "core show channel" output. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields ... This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	289cfe2b4e	Merged revisions 291075 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291075 | rmudgett | 2010-10-11 11:42:54 -0500 (Mon, 11 Oct 2010) | 22 lines Merged revisions 291073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010) | 15 lines Fixed infinite loop in verbose/debug message output. Setting the module/filename specific message level and then changing it resulted in the linked list being looped on itself. Traversing this linked list is an infinite loop if what you are looking for is not in the list. Also plugged some CLI parsing holes in the associated CLI command: * Removing a nonexistent module from the list actually added it with a level of zero. * Setting the non-module specific level to zero is now equivalent to setting it to "off" as documented. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	57535c5989	Merged revisions 282066 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines Add a "core reload" CLI command. Review: https://reviewboard.asterisk.org/r/859/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bbb32fe33e	Merged revisions 282047 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines improved translation paths for wideband codecs The problem I'm addressing is that Asterisk's current method of building the least cost translation paths between codecs does not take into account sample rate. For instance, it was possible for siren14 (a 32khz codec), to contain the a translation path to siren7 (a 16khz audio codec) that goes through slin at 8khz. In this case Asterisk takes a 32khz codec, down samples it to 8khz and then up samples it to 16khz which is terrible regardless if it is computationally less expensive. This patch now builds translation paths that give priority to maintaining the best possible sample rate before taking into consideration computational cost. This patch also adds cli commands to expose what translation paths are actually being used. Changes: 1. Translation paths will never contain a step that changes the sample rate unless absolutely necessary. 2. When choosing the best codec to make two channels compatible. Shared codecs with the highest sample rate are given priority. 3. A new cli command to show all translation paths available for a specific codec 'core show translation paths [codec name]' has been added. 4. 'core show translation' which displays the translation matrix now includes the new higher bit audio codecs in the table. 5. 'core show channel [channel name]' now displays the translation paths if translation is used. (closes issue #16841) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/842/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	e287e4090c	Merged revisions 281529 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281529 | russell | 2010-08-10 11:21:58 -0500 (Tue, 10 Aug 2010) | 8 lines Resolve a problem with channel name tab completion. Hitting tab without typing any part of a channel name resulted in no results. This now results in getting a full list of active channels, just as it did in previous versions of Asterisk. Review: https://reviewboard.asterisk.org/r/818/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring ... The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	98ef8df1ab	Add a CLI command that blocks until Asterisk has fully booted. ... Review: https://reviewboard.asterisk.org/r/684/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	e3c9e6ae86	Fix compilation problem with previous commit. ... (issue #16009) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	a8a1961be7	Add ability to hangup all channels from the CLI. ... Added the keyword 'all' to the 'channel hangup request' CLI command so that you can request all channels to be hungup without having to restart Asterisk. (closes issue #16009) Reported by: moy Patches: hangup-all-rev-221688.patch uploaded by moy (license 222) Tested by: moy, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	7d53dc86d6	Notify CLI when modules is loaded / unloaded ... (closes issue #17308) Reported by: pabelanger Patches: cli.modules.patch uploaded by pabelanger (license 224) Tested by: pabelanger, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ae6008ef3a	Change per-file debug and verbose levels to be per-module, the way ... users expect them to work. 'core set debug' and 'core set verbose' can optionally change the level for a specific filename; however, this is actually for a specific source file name, not the module that source file is included in. With examples like chan_sip, chan_iax2, chan_misdn and others consisting of multiple source files, this will not lead to the behavior that users expect. If they want to set the debug level for chan_sip, they want it set for all of chan_sip, and not to have to also set it for reqresp_parser and other files that comprise the chan_sip module. This patch changes this functionality to be module-name based instead of file-name based. To make this work, some Makefile modifications were required to ensure that the AST_MODULE definition is present in each object file produced for each module as well. Review: https://reviewboard.asterisk.org/r/574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	1a219ad725	Fix compiling with LOW_MEMORY. ... Modified handle_verbose to be LOW_MEMORY aware, removed old RTP related code in chan_sip. (closes issue #16381) Reported by: michael_iedema Patches: ast_complete_source_filename.patch uploaded by michael iedema (license 942) modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	cdd1f9e296	Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it. ... This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the case where multiple results need to be returned; OBJ_NODATA mode already was supported). In addition, it converts ast_channel_iterators (only the targeted versions, not the ones that iterate over all channels) to use this method. During this work, I removed the 'ao2_flags' arguments to the ast_channel_iterator constructor functions; there were no uses of that argument yet, there is only one possible flag to pass, and it made the iterators less 'opaque'. If at some point in the future someone really needs an ast_channel_iterator that does not lock the container, we can provide constructor(s) for that purpose. Review: https://reviewboard.asterisk.org/r/379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	642bec4d6f	AST-2009-005 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	126bf8eeb5	add default alias reload to run module reload. ... Requiring 'module reload' to reload everything, including core etc makes russell very unhappy. The default configuration already loads the 'friendly' aliases template. Added 'reload=module reload' to that template. Also removed the comment in main/cli.c that reload should come back. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d79f6e17fa	Note that "reload" needs to be added back. ... I keep getting annoyed at having to type "module reload" to reload everything, so I'm adding a note that we need to add "reload" back. "module reload" doesn't really make sense as the command to reload everything, including the core. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5d28d72d37	Don't log a warning for something that does not affect operation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. ... CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	ef9b9e36d4	Merged revisions 203380 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines 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/trunk@203381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	3814caadad	Merged revisions 203311 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009) | 2 lines Don't try to free NULL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203338 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
Tilghman Lesher	98ce0e5b6c	Eliminate several needless checks and fix a few memory leaks ... (closes issue #14833) Reported by: contactmayankjain Patches: all_changes.patch uploaded by contactmayankjain (license 740) slightly modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) ... This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. ... There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	11ac1f7e11	Fix building of chan_h323 with gcc-3.3 ... There seems to be a bug with old versions of g++ that doesn't allow a structure member to use the name list. Rename list member to group_list in ast_group_info and change the few places it is used. (closes issue #14790) Reported by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2a4f9f7181	Change global_app_buf to ast_str_thread_global_buf. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	3b08de23df	Remove tabs from comment ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	803a9203d5	Don't crash when typing 'core set verbose' or 'core set debug' by themselves. ... (closes issue #14219) Reported by: jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded by jamesgolovich (license 176) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c8223fc957	Merge ast_str_opaque branch (discontinue usage of ast_str internals) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	c8c8995b70	add tab completion for 'core set debug X filename.c' ... (closes issue #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	1e8e12efcf	Janitor, use ARRAY_LEN() when possible. ... (closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	4900f83f56	Fix minor coding guidelines introduced with CLI permissions. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3d4c0cd421	Merged revisions 160207 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	033bffd32f	Introduce CLI permissions. ... Based on cli_permissions.conf configuration file, we are able to permit or deny cli commands based on some patterns and the local user and group running rasterisk. (Sorry if I missed some of the testers). Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	86f900b201	This commit does two things: ... - Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	805556773f	Fix CLI command 'channel request hangup' ... Prodded on IRC by Russell and fixed by eliel (closes issue #13730) Reported by: eliel Patches: main_cli.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	e235a07376	(closes issue #13557) ... Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Sean Bright	b29fb615ae	Resolve a subtle bug where we would never successfully be able to get ... the first item in the CLI entry list. This was preventing '!' from showing up in either 'help' or in tab completion. (closes issue #13578) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	90751b16ca	Merge the cli_cleanup branch. ... This work is done by lmadsen, junky and mvanbaak during AstriDevCon. This is the second audit the CLI got, and this time lmadsen made sure he had _ALL_ modules loaded that have CLI commands in them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	28764dd1f6	Some fixes to autocompletion in some commands. ... Changes applied by this patch: - Fix autocompletion in 'sip prune realtime', sip peers where never auto completed. Now we complete this command with: 'sip prune realtime peer' -> all | like | sip peers Also I have modified the syntax in the usage, was wrong... - Pass ast_cli_args->argv and ast_cli_args->argc while running autocompletion on CLI commands (CLI_GENERATE). With this we avoid comparisons on ast_cli_args->line like this: strcasestr(a->line, " description") strcasestr(a->line, "descriptions ") strcasestr(a->line, "realtime peer"), and so on.. Making the code more confusing (check the spaces in description!). The only thing we must be sure is to first check a->pos or a->argc. - Fix 'iax2 prune realtime' autocompletion, now we autocomplete this command with 'all' & 'iax2 peers', check a look that iax2 peers where all the peers, now only the ones in the cache.. (closes issue #13133) Reported by: eliel Patches: clichanges.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 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	18 years ago
Sean Bright	3fdc96d0b4	Merged revisions 135597 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug 2008) | 1 line Use PATH_MAX for filenames ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	5b7933fe5e	Janitor patch to change uses of sizeof to ARRAY_LEN ... (closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	4ff527903e	Code wasn't ready to be merged - see -dev list discussion ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	45e79490ba	Implement flags for AGI in the channel structure so taht "show channels" and ... AMI commands can display that a channel is under control of an AGI. Work inspired by work at customer site, but paid for by Edvina AB git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago