summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2011-12-14Fix accidental use of tabs instead of spaces from previous ↵Jonathan Rose
features.conf.sample change ........ Merged revisions 348157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348158 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-14Document PARKINGSLOT variable in features.conf.sampleJonathan Rose
(issue ASTERISK-16239) ........ Merged revisions 348154 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348155 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-13Fix FollowMe CallerID on outgoing calls.Richard Mudgett
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 ........ Merged revisions 348102 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-13Fix possible misshandling of an incoming SIP response as a peer poke response.Stefan Schmidt
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 ........ Merged revisions 348056 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348061 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
2011-12-12Update sample configs to put incoming calls into context public.Richard Mudgett
* 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/ ........ Merged revisions 347953 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-09Adds MixMonitor and StopMixMonitor AMI commands to the managerJonathan Rose
These commands work much like the dialplan applications that would otherwise invoke them. A nice benefit of these is that they can be invoked on a call remotely and at any time during a call. They work much like the Monitor and StopMonitor ami commands. (closes issue ASTERISK-17726) Reported by: Sergio González Martín Patches: mixmonitor_actions.diff uploaded by Sergio González Martín (license 5644) Review: https://reviewboard.asterisk.org/r/1193/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347903 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
2011-12-09Fix some parsing issues in add_exten_to_pattern_tree().Richard Mudgett
* 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 ........ Merged revisions 347812 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-08Fix regression when using tcpenable=no and tlsenable=yes.Walter Doekes
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 ........ Merged revisions 347727 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-08Fix regressed behavior of queue set penalty to work without specifying 'in ↵Jonathan Rose
<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/ ........ Merged revisions 347656 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-08Mark channel running the h exten with the soft-hangup flag.Richard Mudgett
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 ........ Merged revisions 347600 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-08Don't crash on INFO automon request with no channelTerry Wilson
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 ........ Merged revisions 347532 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-08Fix segfault on answer.Damien Wedhorn
Fix a segfault if an attempt to answer a call is made between when the inbound call gives up (and the channel is removed) and when the device is notified and removes the call from the device. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-07Update AMI Getvar and Setvar documentation about supplying a channel name.Richard Mudgett
(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 ........ Merged revisions 347439 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-07Fix: Meetme recording variables from realtime DB use null entries over ↵Jonathan Rose
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 ........ Merged revisions 347383 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347395 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
2011-12-06Make SIP INFO messages for dtmf-relay signals case insensitive.Richard Mudgett
(closes issue ASTERISK-18924) Reported by: Kevin Taylor ........ Merged revisions 347292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347293 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06Documents CHANNEL(musicclass) taking priority over m([x]) in waitExtenJonathan Rose
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 ........ Merged revisions 347240 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06Add VM_INFO() dialplan function to gather information about a mailbox.Walter Doekes
Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname, language, locale, pager, password, tz. (closes issue ASTERISK-18634) Patch by: Kris Shaw Review: https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06Don't allow transport=tcp when tcpenable=no.Walter Doekes
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 ........ Merged revisions 347167 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06Add regression tests for issue ASTERISK-18838.Walter Doekes
Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan ........ Merged revisions 347131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347146 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-06The voicemail [general] zonetag and locale variables weren't loadedWalter Doekes
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 ........ Merged revisions 347124 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347157 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
2011-12-06Fixed crash from orphaned MWI subscriptions in chan_sipMatthew Jordan
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 ........ Merged revisions 347068 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-05Restore call progress code for analog ports.Richard Mudgett
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 ........ Merged revisions 347007 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-05Resolve duplicate label used in multiple priorities for the same extension.Jonathan Rose
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 ........ Merged revisions 346955 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-05Fix chan_jingle/gtalk load regression introduced in r346087Kinsey Moore
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 ........ Merged revisions 346952 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-04For SIP REGISTER fix domain-only URIs and domain ACL bypass.Walter Doekes
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 ........ Merged revisions 346900 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-02Update SIP MESSAGE To parsing to correctly handle URIMatthew Jordan
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/ ........ Merged revisions 346856 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-02implement nat option for rtp channels with ooh323Alexandr Anikin
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-02Merged revisions 346763 via svnmerge from Alexandr Anikin
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines 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/trunk@346777 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
2011-12-01Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a ↵Jonathan Rose
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 ........ Merged revisions 346698 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30Remove the few places where we try to ast_verbose() without a newline.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-30Fix edge case for overflow buffer.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346617 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
2011-11-30Update queues.conf.sample documentation.Leif Madsen
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 ........ Merged revisions 346473 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-29Fix compilation of utilities (caught by Bamboo).Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346429 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
2011-11-28Fix regression that 'rtp/rtcp set debup ip' only works when also a port was ↵Stefan Schmidt
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 ........ Merged revisions 346293 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346294 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
2011-11-23Minor cleanup in chan_sip get_msg_text() function.Walter Doekes
In r116240, get_msg_text() got an extra parameter to fix the unwanted addition of trailing newlines to SIP MESSAGE bodies. This caused all linefeeds to be trimmed, which isn't right either. This is a stop-gap; the right fix is to return the original SIP request body. Review: https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan ........ Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346198 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346199 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
2011-11-23Fixed SendMessage stripping extension from To: header in SIP MESSAGEMatthew Jordan
When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ ........ Merged revisions 346040 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-23Resume playing existing hold music for cached realtime MOHTerry Wilson
As a result of the fix for ASTERISK-18039, realtime caching MOH no longer properly resumes playing back a file between different holds in the same call. This is because scanning for new files causes the existing file array to be emptied and we were just comparing that the saved pointer to the filename matched the pointer to the filename in a particular position in the array. An easy fix is to save the filename instead of a pointer to it and then do a strcmp instead of comparing the addresses. (closes issue ASTERISK-18912) Review: https://reviewboard.asterisk.org/r/1596/ ........ Merged revisions 346030 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346031 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-07 16:20:55 +0000