22559 Commits (a08f48089c6df7169f59aa632b0e92dc978cd42a)

 

All Branches   

Search

Author	SHA1	Message	Date
Sean Bright	c11151084a	Use ast_copy_string instead of strncpy to guarantee a NUL terminated string. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@374132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	a9b6d01456	res_jabber: Remove CLI command 'jabber test' ... The opinion of development was that it is both improper to have Matt's personal email address used in the source and that the command wouldn't be useful without it. (closes issue AST-467) Reported by: Malcolm Davenport ........ Merged revisions 374032 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@374045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	3aaf2c9d21	Update documentation to make it explicit that "stream file" will not restart musiconhold. ... (issue ASTERISK-17367) Reported by: oej ........ Merged revisions 373989 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	efb9b8eabc	Fix SendDTMF crash and channel reference leak using channel name parameter. ... The SendDTMF channel name parameter has two issues. 1) Crashes if the channel name does not exist. 2) Leaks a channel reference if the channel is the current channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF documentation. * Renamed app to senddtmf_name and tweaked the type. ........ Merged revisions 373945 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e1a89a51d2	loader: Ensure dependent modules are properly initialized. ... If an Asterisk module specifies a dependency in ast_module_info.nonoptreq, it is possible for Asterisk to skip calling the modules's .load function. Asterisk was loading and linking the module via load_dynamic_module() but was not adding the module to the resource_heap. Therefore the module was not initialized based on it's priority along with the other modules in the heap. Now use load_resource() instead of load_dynamic_module() for non-optional requirement. This will add the module to the resource_heap so the module can be properly initialized in the correct order. This is required if there are any module global data structures initialized in the .load() callback for the module on platforms which do not support weak references. (issue ASTERISK-20439) Reported by: sruffell Patches: 0001-loader-Ensure-dependent-modules-are-properly-initial.patch uploaded by sruffell (license 5417) ........ Merged revisions 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f9562893c2	Fix an issue where Local channels dialed by app_queue are considered in use immediately. ... The chan_local channel driver returns a device state of in use even if a created Local channel has not yet been dialed. This fix changes the logic to return a state of not in use until the channel itself has been dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach Review: https://reviewboard.asterisk.org/r/2116/ ........ Merged revisions 373878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	10a059f58a	Move handling of 408 response so there is no misleading warning message. ... (closes issue ASTERISK-20060) Reported by: Walter Doekes ........ Merged revisions 373848 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373849 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3265a1f06c	Fixed meetme tab completion and command documentation. ... * Removed unnecessary case sensitivity in meetme list, lock, unlock, mute, unmute, and kick commands. * Separated meetme lock/unlock, mute/unmute, and kick commands into their own registered commands to simplify tab completion and parameter checking. meetme_lock_cmd(), meetme_mute_cmd(), and meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue AST-1006) Reported by: John Bigelow Tested by: rmudgett ........ Merged revisions 373815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	ca280821c7	Fix saying of date in Dutch. ... The Dutch say the date before the month. (closes issue ASTERISK-20353) Reported by: Teun Ouwehand ........ Merged revisions 373773 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f9c03bd789	Remove dead code and documentation for nonexistent feature. ... multiplelogin was removed from chan_agent back in 1.6.0 when AgentCallbackLogin() was removed. (closes issue AST-948) reported by Steve Pitts ........ Merged revisions 373768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	eae21c1a00	Fix error where improper IMAP greetings would be deleted. ... (closes issue ASTERISK-20435) Reported by: fhackenberger Patches: asterisk-20435-imap-del-greeting.diff uploaded by Michael L. Young (License #5026) (with suggested modification made by me) ........ Merged revisions 373735 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373737 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	a949e07a71	Fix T.38 support when used with chan_local in between. ... Users of the T.38 API can indicate AST_T38_REQUEST_PARMS on a channel to request that the channel indicate a T.38 negotiation with the parameters present on the channel. The return value of this indication is expected to be AST_T38_REQUEST_PARMS upon success but with chan_local involved this could never occur. This fix changes chan_local to always return AST_T38_REQUEST_PARMS for this situation. If the underlying channel technology on the other side does not support T.38 this would have been determined ahead of time using ast_channel_get_t38_state and an indication would not occur. (closes issue ASTERISK-20229) Reported by: wdoekes Patches: ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review: https://reviewboard.asterisk.org/r/2070/ ........ Merged revisions 373705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	fbed3175d8	Fix an issue where media would not flow for situations where the legacy STUN code is in use. ... The STUN packets should *not* be blocked by strict RTP. (closes issue ASTERISK-20415) Reported by: Michele Cicciotti patches: uploaded by Joshua Colp (trunk r369817) ........ Merged revisions 373702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	27eb74df81	"show" completion option for "queue" shouldn't appear twice ... When tab-completing CLI commands starting with "queue", "show" appeared twice in the list due to the way that Asterisk's tab completion functions and the order in which the commands were registered. The registration order has been altered to resolve this issue. (closes issue AST-940) Reported-by: Steve Pitts ........ Merged revisions 373666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	aab2448922	Properly handle UAC/UAS roles for SIP session timers ... The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder. This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties. (closes issue AST-922) Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged revisions 373652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	0359dd6391	Fix valgrind found memcpy issues in codec_ilbc. ... Valgrind found codec_ilbc using memcpy instead of memmove for overlapping memory blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231) Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license #5674) patch uploaded by Walter Doekes ........ Merged revisions 373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373645 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fee7a72dd0	Make rebuild GSM, ilbc, or lpc10 codecs if the respective sources change. ... ........ Merged revisions 373618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	45e6f05957	chan_sip: Set Quality of Service for video rtp instance ... (closes issue ASTERISK-20201) Reported by: ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license 6008) ........ Merged revisions 373617 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	0f64dfacec	Fix documentation for default username in res_odbc ... This was previously stated to be "root", but is actually the name of the context if unspecified. (closes issue ASTERISK-20258) Reported by: Stefan x ........ Merged revisions 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	bf976708de	Fix an issue where a caller to ast_write on a MulticastRTP channel would determine it failed when in reality it did not. ... When sending RTP packets via multicast the amount of data sent is stored in a variable and returned from the write function. This is incorrect as any non-zero value returned is considered a failure while a return value of 0 is success. For callers (such as ast_streamfile) that checked the return value they would have considered it a failure when in reality nothing went wrong and it was actually a success. The write function for the multicast RTP engine now returns -1 on failure and 0 on success, as it should. (closes issue ASTERISK-17254) Reported by: wybecom ........ Merged revisions 373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	6e49e5e0f0	Add missing checks that I neglected. The SIP technology and SIP info technology should be considered equal. ... (closes issue ASTERISK-20409) Reported by: michele cicciotti privatewave ........ Merged revisions 373532 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a2e582c835	Revert change to res_rtp_asterisk committed in r373236 (1.8) ... The change committed in r373236 attempted to account for endpoints that increased their RTP timestamp in DTMF end of event re-transmissions. This change attempted to make Asterisk continue to work with endpoints that failed to follow the RFC while maintaining the fix that allowed for out of order DTMF to be handled. Unfortunately, there is no free lunch, and this patch broke any system that sent DTMF immediately after an RTP session was established or when an SSRC is updated. As such, that patch is being reverted for the previous behavior. Endpoints that erroneously increase the RTP timestamp in DTMF end of event packets will not work properly with Asterisk. (issue ASTERISK-20424) ........ Merged revisions 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7ae8943aac	Be consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid> ... When setting CALLERID(pres)=unavailable in the dialplan, the From header in the SIP message contains "Anonymous" <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk should use a lowercase a in the userpart of the URI. * Make the From header use a lowercase A in the userpart of the anonymous URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola ........ Merged revisions 373500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	0d5509fae5	func_audiohookinherit: Document some missed sources. ... This patch also mentions that AUDIOHOOK_INHERIT can be used to transfer MixMonitor audiohooks. There is also wiki that addresses audiohooks and the use of AUDIOHOOK_INHERIT at the following link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........ Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	de75d22add	Fix potential reentrancy problems in chan_sip. ... Asterisk v1.8 and later was not as vulnerable to this issue. * Made find_call() lock each private as it processes the found dialogs. (Primary cause of ABE-2876) * Made the other functions that traverse the dialogs container lock each private as it examines them. * Fix race condition in sip_call() if the thread that sent the INVITE is held up long enough for a response to be processed. The p->initid for the INVITE retransmission could be added after it was canceled by the response processing. * Made __sip_destroy() clean up resource pointers after freeing. This is primarily defensive in case someone has a stale private pointer. * Removed redundant memset() in reqprep(). The call to init_req() already does the memset() and is the first reference to req in reqprep(). * Removed useless set of req.method in transmit_invite(). The calls to initreqprep() and reqprep() have to do this because they memset() the req. JIRA ABE-2876 .......... Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 373424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	612fb43efc	Fix a deadlock caused by a race condition between removing a hint and reloading the dialplan and subscribing to the removed hint. ... If conditions were right it was possible for both the PBX core and chan_sip to deadlock by both having a lock that the other wants. In the case of the PBX core it had the contexts lock and wanted a SIP dialog lock, while in the case of chan_sip it had the SIP dialog lock and wanted the contexts lock. This fix unlocks the SIP dialog before getting the extension state so that the other thread will not block on trying to lock it. Once the extension state is retrieved the SIP dialog is locked again and life carries on. As the SIP dialog is reference counted it is not possible for it to go away after unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins ........ Merged revisions 373438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	a7b98bd2b0	iax2-provision: Fix improper return on failed cache retrieval ... (closes issue ASTERISK-20337) reported by: John Covert Patches: iax2-provision.c.patch uploaded by John Covert (license 5512) ........ Merged revisions 373342 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	443a5cc0e1	app_queue: Make queue reload members and variants of that work ... Prior to this patch, 'queue reload members' cli command did not work at all. This also affects the manager function 'QueueReload' when supplied with the 'members: yes' field. (closes issue AST-956) Reported by: John Bigelow ........ Merged revisions 373298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	2ff2dbe05a	Fix incorrect MeetME conference bridge reference count decrementing and sometimes premature destruction. ... When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one. This operation left around a pointer to the last created conference bridge still containing participants. When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of and the reference count of the conference bridge decremented. If there was only a single participant in the conference bridge it was ultimately destroyed prematurely. (closes issue AST-994) Reported by: John Bigelow ........ Merged revisions 373242 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a57f43ef4e	When processing RFC 2833 DTMF, accomodate increasing timestamps in End events ... While endpoints should not be changing the source timestamp between DTMF event packets, the fact is there exists those endpoints that do exactly that. To work around this, we absorb timestamps within the expected re-transmit period. Note that this period only affects End of Event packets, so it should not prevent the detection of new DTMF digits that happen to arrive right on top of each other. (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson Tested by: mjordan, Vladimir Mikhelson Review: https://reviewboard.asterisk.org/r/2124 ........ Merged revisions 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	2506c8e676	Ensure that all ConfBridge sounds can be set using CONFBRIDGE function ... The CONFBRIDGE function can be used to set the sounds in a ConfBridge bridge profile. Unfortunately, three sounds were missed in the portion of the code that applies the settings passed in from the function: sound_only_one, join, and leave. This patch makes sure that the sounds passed from the function are applied to the bridge profile. (closes issue ASTERISK-20404) Reported by: univ Tested by: mjordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	7af34e67ad	Fix a regression where direct media was not permitted for calls using SIP INFO DTMF. ... A change was committed to fix direct media ACL support. This change wrongly assumed that only a single channel technology structure exists for chan_sip. This is in fact false as a second exists for calls using SIP INFO DTMF. The code which performs direct media ACL checking now checks for both the non-INFO DTMF and INFO DTMF channel technology structures. (closes issue ASTERISK-20409) Reported by: michele cicciotti privatewave ........ Merged revisions 373165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	98e31386fa	Don't crash when passing a NULL message to __astman_get_header. ... Before this commit, __astman_get_header would blindly dereference the passed in 'struct message *' to traverse the header list. There are cases, however, such as '*CLI> sip qualify peer foo' where the message pointer is NULL, so we need to check for that. ........ Merged revisions 373131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	198b9dab95	Made companding law for SS7 calls only determined by SS7 signaling type. ... For SS7, the companding law for a call was chosen inconsistently depending upon ss7type (ITU vs ANSI) and the DAHDI companding default (T1 vs E1). For incoming calls, the companding law was determined by ss7type. For outgoing calls, the companding law was determined by the DAHDI default. With the wrong combination you would get A-law/u-law conflicts. An A-law/u-law conflict sounds like bad static on the line. SS7 ITU signaling with E1 line: ok SS7 ITU signaling with T1 line: noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling with T1 line: ok * Fix the companding law used to be determined by the SS7 signaling type only. ........ Merged revisions 373090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d0844c54d4	Resolve memory leaks in TLS initialization and TLS client connections ... This patch resolves two sources of memory leaks when using TLS in Asterisk: 1) It removes improper initialization (and multiple re-initializations) of portions of the SSL library. Asterisk calls SSL_library_init and SSL_load_error_strings during SSL initialization; collectively this obviates the need for calling any of the following during initialization or client connection handling: * ERR_load_crypto_strings (handled by SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for SSL_library_init) 2) Failure to completely clean up all memory allocated by Asterisk and by the SSL library for TLS clients. This included not freeing the SSL_CTX object in the SIP channel driver, as well as not clearing the error stack when the TLS client exited. Note that these memory leaks were found by Thomas Arimont, and this patch was essentially written by him with some minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas Arimont (license 5525) Review: https://reviewboard.asterisk.org/r/2105 ........ Merged revisions 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1f63923c93	Constify __ao2_ref_debug in astobj2 ... When REF_DEBUG is enabled in certain files - most notably ccss.c - the 'tag' parameter passed to __ao2_ref_debug will be a const char *. The function currently expects that parameter to not be const. This causes a warning when compiling, as the const qualifier is being discarded. With dev-mode enabled, this prevents compiling Asterisk. This patch makes __ao2_ref_debug's tag and file parameters const. (closes issue ASTERISK-20408) Reported by: mjordan ........ Merged revisions 372959 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	ba79960450	Fix timeouts for ast_waitfordigit[_full]. ... ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds, expecting it to decrement the timeout by however many milliseconds were waited. This is a problem if it consistently waits less than 1ms. The timeout will never be decremented, and we wait... FOREVER! This patch makes ast_waitfordigit_full manage the timeout itself. It maintains the previously undocumented behavior that negative timeouts wait forever. (closes issue ASTERISK-20375) Reported by: Mark Michelson Tested by: Mark Michelson Review: https://reviewboard.asterisk.org/r/2109/ ........ Merged revisions 373024 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@373025 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a9991552d0	Add channel name to a warning to make debugging easier. ... The "autodestruct with owner in place" message is typically indicative of a channel reference leak. Printing out the name of the channel in the message may be helpful when trying to debug the issue. ........ Merged revisions 372932 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	81e064cc09	chan_local: Switch from using a random 4 digit hex identifier to unique id ... Changes chan_local channels to use an 8 digit hex identifier generated atomically and sequentially in order to eliminate the chance of having multiple channels with the same name during high call volume situations. (issue ASTERISK-20318) Reported by: Dan Cropp Review: https://reviewboard.asterisk.org/r/2104/ ........ Merged revisions 372902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9fd24fa894	Fix inability to shutdown gracefully due to an unending channel reference. ... message.c makes use of a special message queue channel that exists in thread storage. This channel never goes away due to the fact that the taskprocessor used by message.c does not get shut down, meaning that it never ends the thread that stores the channel. This patch fixes the problem by shutting down the taskprocessor when Asterisk is shut down. In addition, the thread storage has a destructor that will release the channel reference when the taskprocessor is destroyed. (closes issue AST-937) Reported by Jason Parker Patches: AST-937.patch uploaded by Mark Michelson (License #5049) Tested by Jason Parker git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	265ac96d58	Corrects the astsbindir setting when installing the sample asterisk.conf. ... (closes issue ASTERISK-20406) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	c7582c9da8	Fix bad channel application data reference. ... When channels get bridged due to an AMI bridge action or a DTMF attended transfer, the two channels that get bridged have their application data pointing to the other channel's name. This means that if one channel is hung up but the other moves on, it means that the channel that moves on will have its application data pointing at freed memory. (issue ASTERISK-20335) Reported by: aragon ........ Merged revisions 372840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	537d53dba5	Ensure iax2 debug output is displayed when expected ... When IAX2 debug was changed from iax_showframe to iax_outputframe, some instances were missed (or added afterward). This was causing debug output to not be displayed when expected. (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by: John Covert ........ Merged revisions 372804 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	7df4087529	app_meetme: Document that 'p' option will continue in dialplan. ... (closes issue AST-991) Reported by John Bigelow ........ Merged revisions 372765 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	f841f9843e	Warn on CLI when UDPTL init fails ... This adds a CLI warning when a SDP offer is rejected due to UDPTL initialization failure. Previously, there was no indication of the reason for offer rejection in this case. (closes issue ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi ........ Merged revisions 372763 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	9e6febf229	Masquerade: Retain parkinglot settings made by CHANNEL function. ... Prior to this patch, the user would have a parkinglot set on a channel that was parked and when the channel was retrieved, any attempt by that channel to park would simply use the default. This patch makes parkinglot values set in this way be retained through the masquerade. (closes issue AST-990) Reported by: Nick Huskinson Patches: masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182) ........ Merged revisions 372736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372737 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ac624508eb	Only re-create an SRTP session when needed ... In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an SDP offer and the ability to re-create an SRTP session when the crypto keys changed. In certain circumstances - most notably when a phone is put on hold after having been bridged for a significant amount of time - the act of re-creating the SRTP session causes problems for certain models of phones. The patch committed in r356604 always re-created the SRTP session regardless of whether or not the cryptographic keys changed. Since this is technically not necessary, this patch modifies the behavior to only re-create the SRTP session if Asterisk detects that the remote key has changed. This allows models of phones that do not handle the SRTP session changing to continue to work, while also providing the behavior needed for those phones that do re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported by: Nicolo Mazzon Tested by: Nicolo Mazzon Review: https://reviewboard.asterisk.org/r/2099 ........ Merged revisions 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	8c831bcb52	Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c. ... Without this flag, those files will compile with the system installed OpenSSL headers (if they exist). This is a real bummer if a different path was specified using --with-ssl= (closes issue ASTERISK-20392) ........ Merged revisions 372682 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	12a2affd6f	Fix MALLOC_DEBUG version of ast_strndup(). ... (closes issue ASTERISK-20349) Reported by: Brent Eagles ........ Merged revisions 372655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9a2a1108cf	Remove annoying unconditional debug message from INC/DEC functions. ... (closes issue AST-1001) Reported by: Guenther Kelleter ........ Merged revisions 372628 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@372629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago