asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	d43b217d08	Fix extension state callback references in chan_sip. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	8e42bdbc7d	Fix for memory leaks / cleanup in cel_pgsql There were a number of issues in cel_pgsql's pgsql_log method: * If either sql or sql2 could not be allocated, the method would return while the pgsql_lock was still locked * If the execution of the log statement succeeded, the sql and sql2 structs were never free'd * Reconnection successes were logged as ERRORs. In general, the severity of several logging statements was reduced (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/ ........ Merged revisions 348888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	fe36eef8f1	Add Asterisk TestSuite event hooks to support ConfBridge testing This patch adds initial testsuite event hooks so that ConfBridge tests can be executed in the Asterisk TestSuite. (issue ASTERISK-19059) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	07fd1458d7	Allow packetization vaules > 127 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348845 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8ac80163cc	Make codecs/speex ignore *.i files also. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d11c058791	Make apps/confbridge ignore *.i files also. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5bfbc7febf	Fix chan_iax2 to not report an RDNIS number if it is blank. Some ISDN switches complain or block the call if the RDNIS number is empty. * Made chan_iax2 not save a RDNIS number into the ast_channel if the string is blank. This is what other channel drivers do. (closes issue ASTERISK-17152) Reported by: rmudgett ........ Merged revisions 348735 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d351aa67ab	Fix crashes on other platforms caused by interference from Darwin weak symbol support. Support weak symbols on a platform specific basis. The Mac OS X (Darwin) support must be isolated from the other platforms because it has caused other platforms to crash. Several other platforms including Linux have GCC versions that define the weak attribute. However, this attribute is only setup for use in the code by Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged revisions 348647 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	0abad2d894	Update documentation for MESSAGE_SEND_STATUS variable. (Closes issue ASTERISK-19056) Reported by: Yuri Patches: 348360.diff uploaded by Yuri (license #5242) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	f501072849	Correct two flaws in sip.conf.sample related to AST-2011-013. * The sample file listed two values for the 'nat' option as being the default. Only 'force_rport' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. ........ Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	045326d869	Clean-up on isle five for __ast_request_and_dial() and ast_call_forward(). * Add locking when a channel inherits variables and datastores in __ast_request_and_dial() and ast_call_forward(). Note: The involved channels are not active so there was minimal potential for problems. * Remove calls to ast_set_callerid() in __ast_request_and_dial() and ast_call_forward() because the set information is for the wrong direction. * Don't use C++ keywords for variable names in ast_call_forward(). * Run the redirecting interception macro if defined when forwarding a call in ast_call_forward(). Note: Currently will never execute because the only callers that supply a calling channel supply a hungup or zombie channel. * Make feature_request_and_dial() put the transferee into autoservice when it calls ast_call_forward() in case a redirection interception macro is run. Note: Currently will never happen because the caller channel (Party B) is always hungup at this time. * Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame to silence a log message. ........ Merged revisions 348464 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e95b1dc220	Fix cut and past error in ast_call_forward(). (issue ASTERISK-18836) ........ Merged revisions 348401 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5688637636	Fix crash during CDR update. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	09fab9c14a	Fix ParkAndAnnounce to pass the CallerID to the announcing channel. ParkAndAnnounce tried to pass the CallerID to the announcing channel but the ID was wiped out by the channel masquerade done when parking the call. * Save the CallerID before parking the channel to pass it to the announcing channel. * Fixed a minor memory leak in ParkAndAnnounce. * Updated some ParkAndAnnounce log messages. ........ Merged revisions 348310 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a3ac30d14c	Added support for all slin formats to app_originate Previously, app_originate could not originate a call into a non-8kHz conference bridge as the formats for non-8kHz slin codecs were not applied to the created channel. This patch adds all of the formats by default, such that if a created channel has a codec that supports a higher sampling rate, a translation path can be built between it and other channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	d4345a0683	Don't clear LOCALSTATIONID before sending or receiving. The user may set that variable. ASTERISK-18921 ........ Merged revisions 348212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	3d75189139	Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has two required parameters (queuename, option). It was only checking for the presence of queuename. The patch checks for the existence of the option parameter and provides better error logging when invalid values are provided for the option parameter as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1820e23330	Fix accidental use of tabs instead of spaces from previous features.conf.sample change ........ Merged revisions 348157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	f055e5602b	Document PARKINGSLOT variable in features.conf.sample (issue ASTERISK-16239) ........ Merged revisions 348154 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f71c13ed3d	Fix FollowMe CallerID on outgoing calls. The addition of the Connected Line support changed how CallerID is passed to outgoing calls. The FollowMe application was not updated to pass CallerID to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. * Restructured findmeexec() to fix several memory leaks and eliminate some duplicated code. * Made check the return value of create_followme_number(). Putting a NULL into the numbers list is bad if create_followme_number() fails. * Fixed a couple uses of ast_strdupa() inside loops. * The changes to bridge_builtin_features.c fix a similar CallerID issue with the bridging API attended and blind transfers. (Not used at this time.) (closes issue ASTERISK-17557) Reported by: hamlet505a Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1612/ ........ Merged revisions 348101 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	b81ed5a885	Fix possible misshandling of an incoming SIP response as a peer poke response. Also make sure peer has even qualify enabled when handle a peer poke response. (closes issue ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed by: David Vossel ........ Merged revisions 348048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	176819727b	Add a separate buffer for SRTCP packets The function ast_srtp_protect used a common buffer for both SRTP and SRTCP packets. Since this function can be called from multiple threads for the same SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the packets to become corrupted as the buffer was used by both threads simultaneously. This patch adds a separate buffer for SRTCP packets to avoid the problem. (closes issue ASTERISK-18889, Reported/patch by Daniel Collins) ........ Merged revisions 347995 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7af870fd2f	Reverting -r347953 for ASTERISK-14122 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	54f07e2b28	Update sample configs to put incoming calls into context public. * Add warning about the SIP allowguest option in context public. (closes issue ASTERISK-14122) Reported by: Alec Davis Review: https://reviewboard.asterisk.org/r/719/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e9ba4938d1	Fix some parsing issues in add_exten_to_pattern_tree(). * Simplify compare_char() and avoid potential sign extension issue. * Fix infinite loop in add_exten_to_pattern_tree() handling of character set escape handling. * Added buffer overflow checks in add_exten_to_pattern_tree() character set collection. * Made ignore empty character sets. * Added escape character handling to end-of-range character in character sets. This has a slight change in behavior if the end-of-range character is an escape character. You must now escape it. * Fix potential sign extension issue when expanding character set ranges. * Made remove duplicated characters from character sets. The duplicate characters lower extension matching priority and prevent duplicate extension detection. * Fix escape character handling when the escape character is trying to escape the end-of-string. We could have continued processing characters after the end of the exten string. We could have added the previous character to the pattern matching tree incorrectly. (closes issue ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	7af6cc7507	Fix regression when using tcpenable=no and tlsenable=yes. The tlsenable settings are tucked away in main/tcptls.c, so I missed them when resolving ASTERISK-18837. This should resolve the test suite breakage of the sip tls tests. Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt Jordan ........ Merged revisions 347718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1ebeb108c8	Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>' r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when attempting to set penalty on an interface without specifying a queue in the queue set penalty CLI command. In addition, no attempt would be made whatsoever to perform the penalty setting on all the queues in the core list with either the cli command or the non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to document and rename some functions required by this command to better represent what they actually do. Oh yeah, and the use of this command without specifying a specific queue actually works now. Review: https://reviewboard.asterisk.org/r/1609/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	132cbead53	Mark channel running the h exten with the soft-hangup flag. When a bridge is broken, ast_bridge_call() might execute the h exten on the calling channel. However, that channel may not have been the channel that broke the bridge by hanging up. The channel executing the h exten must be in a hung up state so things like AGI run in the correct mode. * Make sure ast_bridge_call() marks the channel it is executing the h exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to match the pbx.c main dialplan execution loop when it executes the h exten.) (closes issue ASTERISK-18811) Reported by: David Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: David Hajek, rmudgett ........ Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b20f27d3f0	Don't crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. (closes issue ASTERISK-18805) ........ Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	31f37c13b1	Update AMI Getvar and Setvar documentation about supplying a channel name. (closes issue ASTERISK-18958) Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ec1a9f4518	Fix: Meetme recording variables from realtime DB use null entries over channel variables Meetme would attempt to substitute the realtime values of RECORDING_FILE and RECORDING_FORMAT from the meetme db entry instead of using the channel variable set for those variables in spite of those database entries being NULL or even lacking a column to represent them. (closes issue ASTERISK-18873) Reported by: Byron Clark Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157) ........ Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	309c50e4bb	Add ASTSBINDIR to the list of configurable paths 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	411fc37ea7	Make SIP INFO messages for dtmf-relay signals case insensitive. (closes issue ASTERISK-18924) Reported by: Kevin Taylor ........ Merged revisions 347292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	0e2b51ac20	Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten If waitExten specifies a music class to use with its music on hold option, it will use CHANNEL(musicclass) instead if that channel variable has been set on the initiating channel. This documents that behavior in the waitExten app so that this can be known without checking the documentation of the code in function local_ast_moh_start. (closes issue ASTERISK-18804) ........ Merged revisions 347239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	7b69c9ee28	Don't allow transport=tcp when tcpenable=no. When tcpenable=no, sending to transport=tcp hosts was still allowed. Resolving the source address wasn't possible and yielded the string "(null)" in SIP messages. Fixed that and a couple of not-so-correct log messages. (closes issue ASTERISK-18837) Reported by: Andreas Topp Review: https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan ........ Merged revisions 347166 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	585499074d	Add regression tests for issue ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan ........ Merged revisions 347131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	2ebbe7c31d	Move setting of voicemail zonetag and locale up a bit. The voicemail [general] zonetag and locale variables weren't loaded until after the mailboxes were initialized. This caused the settings to be unset for those mailboxes until a reload was performed. (closes issue ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570 Reviewed by: Matt Jordan ........ Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	d539515844	Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Reported by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches: blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283) Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged revisions 347058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1e12d2aee4	Restore call progress code for analog ports. Extracting sig_analog from chan_dahdi lost call progress detection functionality. * Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patches: chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.h.diff (license #5685) patch uploaded by Richard Miller ........ Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	fd5420c10e	Resolve duplicate label used in multiple priorities for the same extension. Prior to this patch, if labels with the same name were used for different priorities in the same extension, the new label would be accepted, but it would be unusable since attempts to reach that label would just go to the first one. Now pbx.c detects this, generates a warning in logs, and culls the label before adding it to the dialplan. (closes issue ASTERISK-18807) Reported by: Kenneth Shumard Patches: pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........ Merged revisions 346954 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	e47bf3cf2b	Fix chan_jingle/gtalk load regression introduced in r346087 Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland ........ Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	51c8db4fb1	For SIP REGISTER fix domain-only URIs and domain ACL bypass. The code that allowed admins to create users with domain-only uri's had stopped to work in 1.8 because of the reqresp parser rewrites. This is fixed now: if you have a [mydomain.com] sip user, you can register with useraddr sip:mydomain.com. Note that in that case -- if you're using domain ACLs (a configured domain list) -- mydomain.com must be in the allow list as well. Reviewboard r1606 shows a list of registration combinations and which SIP response codes are returned. Review: https://reviewboard.asterisk.org/r/1533/ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes issue ASTERISK-18741) ........ Merged revisions 346899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	058a8177e4	Update SIP MESSAGE To parsing to correctly handle URI The previous patch (r346040) incorrectly parsed the URI in the presence of a port, e.g., user@hostname:port would fail as the port would be double appended to the SIP message. This patch uses the parse_uri function to correctly parse the URI into its username and hostname parts, and places them in the correct fields in the sip_pvt structure. (issue ASTERISK-18903) Review: https://reviewboard.asterisk.org/r/1597/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	2ff2906011	Merged revisions 346762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r346762 \| may \| 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) \| 7 lines process null frame pointer returned by ast_rtp_instance_read correctly (closes issue ASTERISK-16697) Reported by: under Patches: segfault.diff (License #5871) patch uploaded by under ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ae9145e400	Re-resolve the STUN address if a STUN poll fails for res_stun_monitor. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	53e3ed3b34	Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing. 183 is actually a session progress message. (closes issue ASTERISK-18925) Reported by: Sebastian Denz Tested by: jrose Patches: asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139) ........ Merged revisions 346697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	e0bcc2b29d	r346525 \| jrose \| 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) \| 18 lines 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	e1049ae52b	Update queues.conf.sample documentation. Update the documentation surrounding the use of MONITOR_EXEC to make it more clear that it can be used for both Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413) Reported by: David Woolley Patches: issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026) ........ Merged revisions 346472 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	5d452c2647	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/branches/10@346349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	4b0486b824	Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified. (closes issue ASTERISK-18693) Reported by: Davide Dal Fra Review: https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter Doekes ........ Merged revisions 346292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@346293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

1 2 3 4 5 ...

21986 Commits (19af5d61033858e7e3560219b346593e2cb27e39) All Branches Search

21986 Commits (19af5d61033858e7e3560219b346593e2cb27e39)

All Branches