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2012-01-24Opaquify channel stringfieldsTerry Wilson
Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-19Eliminates doubling the :port part of SIP Notify Message-Account headers.Jonathan Rose
This patch prevents the domain string from getting mangled during the initreqprep step by moving the initialization to before its immediate use. It also documents this pitfall for the ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported by: Yuri Review: https://reviewboard.asterisk.org/r/1678/ ........ Merged revisions 351559 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351560 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Run bootstrap.sh for the for the ASTERISK-18929 fixKinsey Moore
configure and autoconfig.h.in were not regenerated when the fix was committed. ........ Merged revisions 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350737 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-13Fix crash from bridge channel hangup race condition in ConfBridgeMatthew Jordan
This patch addresses two issues in ConfBridge and the channel bridge layer: 1. It fixes a race condition wherein the bridge channel could be hung up 2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object Patch by David Vossel (mjordan was merely the commit monkey) (issue ASTERISK-18988) (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628) (closes issue ASTERISK-19100) Reported by: Matt Jordan Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1654/ ........ Merged revisions 350550 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Replace direct access to channel name with accessor functionsTerry Wilson
There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-31Constify tag argument in REF_DEBUG related code.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-28Improve T.38 gateway V.21 preamble detection.Kevin P. Fleming
This commit removes the V.21 preamble detection code previously added to the generic DSP implementation in Asterisk, and instead enhances the res_fax module to be able to utilize V.21 preamble detection functionality made available by FAX technology modules. This commit also adds such support to res_fax_spandsp, which uses the Spandsp modem tone detection code to do the V.21 preamble detection. There should be no functional change here, other than much more reliable V.21 preamble detection (and thus T.38 gateway initiation). ........ Merged revisions 349248 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-27Fix timing source dependency issues with MOHMatthew Jordan
Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patches: asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026) Review: https://reviewboard.asterisk.org/r/1578/ ........ Merged revisions 349194 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349195 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-23Fix extension state callback references in chan_sip.Richard Mudgett
Chan_sip gives a dialog reference to the extension state callback and assumes that when ast_extension_state_del() returns, the callback cannot happen anymore. Chan_sip then reduces the dialog reference count associated with the callback. Recent changes (ASTERISK-17760) have resulted in the potential for the callback to happen after ast_extension_state_del() has returned. For chan_sip, this could be very bad because the dialog pointer could have already been destroyed. * Added ast_extension_state_add_destroy() so chan_sip can account for the sip_pvt reference given to the extension state callback when the extension state callback is deleted. * Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy() and handle_statechange() now that the struct ast_state_cb has a destructor to call. * Ensure that ast_extension_state_add_destroy() will never return -1 or 0 for a successful registration. * Fixed pbx.c statecbs_cmp() to compare the correct information. The passed in value to compare is a change_cb function pointer not an object pointer. * Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for deadlocking when those locks are held during the callback. * Removed unused lock declaration for the pbx.c store_hints list. (closes issue ASTERISK-18844) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/1635/ ........ Merged revisions 348940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348952 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-22Allow packetization vaules > 127Terry Wilson
According to the RTP packetization documentation, and the maximum values listed in AST_FORMAT_LIST, we should support values > that the signed char array that ast_codec_pref makes available to store the value. All places in the code treat the framing field as though it were an int array instaead of a char array anyway, so this just fixes the type of the array. (closes issue ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/ ........ Merged revisions 348833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348845 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-16Fix crash during CDR update.Richard Mudgett
The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348363 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-12Backed out core changes from r346391Matthew Jordan
During testing, it was discovered that there were a number of side effects introduced by r346391 and subsequent check-ins related to it (r346429, r346617, and r346655). This included the /main/stdtime/ test 'hanging', as well as the remote console option failing to receive the appropriate output after a period of time. I only backed out the changes to main/ and utils/, as this was adequate to reverse the behavior experienced. (issue ASTERISK-18974) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-09Remove autojump extensions from SayUnixTime, make an option to perform ↵Jonathan Rose
automatic jumps. When a caller sends DTMF while the SayUnixTime application is saying the time, The call would jump to the next extension much like it does during Background(). This patch adds option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch allows arguments to sayunixtime to not be used as empty strings in the case of something like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue ASTERISK-16675) Reported by: jlpedrosa Patches: patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959) Review: https://reviewboard.asterisk.org/r/956/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-07Add ASTSBINDIR to the list of configurable pathsTerry Wilson
This patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH. (closes issue ASTERISK-18959) Review: https://reviewboard.asterisk.org/r/1613/ ........ Merged revisions 347344 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06Doubly linked lists unit test and update to implementation.Richard Mudgett
Update the doubly linked list implementation. Now safe traversing can insert before and after the current node when traversing in either direction. Updated the linked lists unit test test_linkedlist to also test doubly linked lists. The old test_dlinkedlist requires a manual check of results and probably should be removed. Review: https://reviewboard.asterisk.org/r/1569/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-01Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.Richard Mudgett
The STUN socket must remain open between polls or the external address seen by the STUN server is likely to change. However, if the STUN request poll fails then the STUN server address needs to be re-resolved and the STUN socket needs to be closed and reopened. * Re-resolve the STUN server address and create a new socket if the STUN request poll fails. * Fix ast_stun_request() return value consistency. * Fix ast_stun_request() to check the received packet for expected message type and transaction ID. * Fix ast_stun_request() to read packets until timeout or an associated response packet is found. The stun_purge_socket() hack is no longer required. * Reduce ast_stun_request() error messages to debug output. * No longer pass in the destination address to ast_stun_request() if the socket is already bound or connected to the destination. (closes issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1595/ ........ Merged revisions 346700 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346701 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 linesJonathan Rose
Cleaning up chan_sip/tcptls file descriptor closing. This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged revisions 346564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346565 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30Reverting 346525 due to accidental patch against trunk instead of 1.8Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30Cleaning up chan_sip/tcptls file descriptor closing.Jonathan Rose
This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-29Allow each logging destination and console to have its own notion of the ↵Tilghman Lesher
verbosity level. Review: https://reviewboard.asterisk.org/r/1599 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-29Merged revisions 346349 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines Fixes memory leak in message API. The ast_msg_get_var function did not properly decrement the ref count of the var it retrieves. The way this is implemented is a bit tricky, as we must decrement the var and then return the var's value. As long as the documentation for the function is followed, this will not result in a dangling pointer as the ast_msg structure owns its own reference to the var while it exists in the var container. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-23Fix calls to ast_get_ip() not initializing the address family.Richard Mudgett
........ Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346240 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-23Fix ast_str_truncate signedness warning and documentation.Walter Doekes
Review: https://reviewboard.asterisk.org/r/1594 ........ Merged revisions 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346145 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-23Fix res_jabber resource leaksKinsey Moore
This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-22Fix dnsmgr entries to ask for the same address family each time.Richard Mudgett
The dnsmgr refresh would always get the first address found regardless of the original address family requested. So if you asked for only IPv4 addresses originally, you might get an IPv6 address on refresh. * Saved the original address family requested by ast_dnsmgr_lookup() to be used when the address is refreshed. ........ Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345977 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-22Clarify why the AST_LOG_* macros exist next to the LOG_* macros.Walter Doekes
(issue ASTERISK-17973) ........ Merged revisions 345923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345924 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-11Use __alignof__ instead of sizeof for stringfield length storage.Walter Doekes
Kevin P Fleming suggested that r343157 should use __alignof__ instead of sizeof. For most systems this won't be an issue, but better fix it now while it's still fresh. Review: https://reviewboard.asterisk.org/r/1573 ........ Merged revisions 344843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344845 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-06Formatting and doxygen improvementsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-02Ensure that string field lengths are properly alignedWalter Doekes
Integers should always be aligned. For some platforms (ARM, SPARC) this is more important than for others. This changeset ensures that the string field string lengths are aligned on *all* platforms, not just on the SPARC for which there was a workaround. It also fixes that the length integer can be resized to 32 bits without problems if needed. (closes issue ASTERISK-17310) Reported by: radael, S Adrian Reviewed by: Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review: https://reviewboard.asterisk.org/r/1549 ........ Merged revisions 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343158 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-01Several fixes to the chan_sip dynamic realtime peer/user lookupWalter Doekes
There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to *still* have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 ........ Merged revisions 342927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342929 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-29Whitespace and some better macro variable names.Richard Mudgett
* Renamed AST_LIST_TRAVERSE_SAFE_BEGIN __new_prev to __list_current. * Renamed AST_LIST_MOVE_CURRENT __list_cur to __extracted. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-29Fix AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable.Richard Mudgett
AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration or before AST_LIST_REMOVE_CURRENT() without corrupting the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed cut and paste error using the wrong variable in AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342662 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-25Return NULL when no results returned for realtime_multientryTerry Wilson
It was not documented what the return value should be when no entries were returned with the multientry realtime callback. This change forces consistent behavior even if the backends return an empty ast_config. Review: https://reviewboard.asterisk.org/r/1521/ ........ Merged revisions 342223 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342224 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-24Fix ao2obj.h comment typos and add missing link/unlink nolock debug defines.Richard Mudgett
........ Merged revisions 342183 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-20Fix AGI exec Park to honor the Park application parameters.Richard Mudgett
The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341718 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-20Clean up ast_check_digitsTerry Wilson
The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. ........ Merged revisions 341529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341530 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-19Don't use is_int() since it doesn't link well on all platformsTerry Wilson
Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341380 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-18More parking issues.Richard Mudgett
* Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-11Update SHA1 code to RFC 6234Tzafrir Cohen
RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-11Convert registered AMI actions to ao2 objects.Richard Mudgett
* Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon ........ Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340281 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-10Merged revisions 340109 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-06Merged revisions 339720 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339720 | rmudgett | 2011-10-06 17:58:40 -0500 (Thu, 06 Oct 2011) | 27 lines Merged revisions 339719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines Fix regression in configure script for libpri capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2 persistence issues with some telcos. ASTERISK-18535 attempted to fix the unexpected requirement that libpri *must* have that feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI optional features required. Unfortunately, I thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and deleted those lines for libpri. The result was the HAVE_PRI_xxx defines that control the ability to use optional libpri features were also deleted. * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional features in a library that the source code could take advantage of if the code supports the feature. (closes issue ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-05Add generic faxdetect framehook to res_faxGregory Nietsky
Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan faxdetect allowing more flexibility. as soon as a fax tone is detected the framehook is removed. there is a penalty involved in running this framehook on non G711 channels as they will be transcoded. CNG tone is suppresed using the SQUELCH flag to allow WaitForNoise to be run on the channel to detect Voice. (Closes issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew Nicholson, Kevin Fleming Review: https://reviewboard.asterisk.org/r/1116/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338253 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338253 | rmudgett | 2011-09-28 16:22:05 -0500 (Wed, 28 Sep 2011) | 14 lines Merged revisions 338235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) | 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch (license #6278) patch uploaded by Luke H ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-26Merged revisions 337974 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337595,337597 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ ........ r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20Merged revisions 337120 via svnmerge from Matthew Jordan
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-19Merged revisions 336734 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336734 | tilghman | 2011-09-19 15:29:40 -0500 (Mon, 19 Sep 2011) | 18 lines Merged revisions 336733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines Various changes to allow 1.8 to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6 extended to work on 10.7 and later. * Now uses the 'weak' symbol for Lion systems, which no longer support 'weak_import' Closes ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-16Merged revisions 336307 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-14Merged revisions 335912 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335912 | rmudgett | 2011-09-14 13:31:15 -0500 (Wed, 14 Sep 2011) | 20 lines Merged revisions 335911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines Remove unnecessary libpri dependency checks in the configure script. Using the --with-pri option with the configure script generated an error about not having PRI_L2_PERSISTENCE if you did not have the absolute latest libpri SVN checkout installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to be for libraries that are dependent upon other libraries and not necessarily for optional/added features within a library. (closes issue ASTERISK-18535) Reported by: Michael Keuter ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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