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2013-01-16Multiple revisions 379209-379210Matthew Jordan
........ r379209 | mjordan | 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines Add module tags to documentation for res_jabber/res_xmpp Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.), the XML documentation for each needs to call out which module is providing the documentation. The module attribute has been added to the various XML fragments for this purpose. ........ r379210 | mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines Update the dtd to actually *support* the module attribute in all elements Mea culpa. ........ Merged revisions 379209-379210 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-16Fix parsing SMSSRC for SMS messagesMatthew Jordan
The parser for SMS messages would incorrectly parse out the from number. The parsing would incorrectly start scanning for the from number at the same index as the first double quote ("); this would inadvertently cause it to treat the first double quote as the terminating double quote for the from number as well. The SMSSRC should now populate correctly. (closes issue ASTERISK-16822) Reported by: menschentier Tested by: Jonas Falck patches: fixSMSSRC.patch uploaded by jonax (license 6320) (closes issue ASTERISK-19153) Reported by: Panos Gkikakis patches: sms-sender-fix.diff uploaded by roeften (license 5884) ........ Merged revisions 379178 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 379179 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-16Set the INVALID_EXTEN channel variable when chan_misdn forces the 'i' extensionMatthew Jordan
The chan_misdn channel driver will send a channel with an invalid destination to the 'i' extension itself if said extension can be reached. It forgot, however, to set the INVALID_EXTEN channel variable when it bounces the channel to this extension. Dialplan writers everywhere moaned at yet another inconsistency. This is yet another example of why duplicating logic in multiple places results in bugs that stick around in Jira for just under three years. Yes: ASTERISK-15456 was created on January 18th, 2010. Patch committed on January 15th, 2013. Ouch. (closes issue ASTERISK-15456) Reported by: Thomas Omerzu patches: chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license 5927) ........ Merged revisions 379145 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 379146 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-15Add busy detection to chan_mobileMatthew Jordan
From the patch author: "First this patch adds general support for busy detection. It also adds support for the ECAM command at Sony Ericsson phones and also signals busy when only early media was received but the call got not answered." Review: https://reviewboard.asterisk.org/r/323 (closes issue ASTERISK-14527) Reported by: Artem Makhutov Tested by: Artem Makhutov patches: busy-full5.patch uploaded by artem (license 5757) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-15Fix ast_bridge_features_register() not registering builtin features. I ↵Richard Mudgett
broke. Ooops. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-14Fixed doc comment for ast_test_validateDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-14Gently reduce masquerade insanityDavid M. Lee
Masquerades are an insane implementation detail within Asterisk. It generates a number of useless and confusing events, and manipulates channels in a way that semantically doesn't make sense. I've given a fairly thorough review of masquerade code and its usage on the wiki at https://wiki.asterisk.org/wiki/x/IwBRAQ. While ultimately it makes the most sense to abandon masquerades altogether, it will take some time to completely irradicate. Even then, there may always be code that's not worth rewriting to get rid of the masquerade. This patch does two things to make masquerades slightly less insane: * When swapping the names of the original and clone channel, only emit a single rename event of original -> original<ZOMBIE>. The original code issued three rename events to accomplish the same end. * In addition to swapping the names of the channels, also swap their uniqueid's. This allows the 'Uniqueid' field to be used as a stable identifier for a channel from and external interface, such as AMI. Review: https://reviewboard.asterisk.org/r/2266/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-14Fix XML encoding of 'identity display' in NOTIFY messages, continued.David M. Lee
When r378933 was merged into 1.8, it should have also escaped remote_display, since it will have the same XML encoding problem when the caller/callee roles are reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter ........ Merged revisions 379001 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 379020 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-13Reset RTP timestamp; sequence number on SSRC changeMatthew Jordan
In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to better account for out of order RTP packets. This was accomplished by using the RTP timestamp and sequence number to check for out of order packets. However, when a SSRC change occurs, the timestamp and sequence number will no longer have any relation to the previously received packets. The variables tracking the timestamp and sequence number therefore have to be reset. (closes issue ASTERISK-20906) Reported by: Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442) ........ Merged revisions 378967 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378984 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-12Fix XML encoding of 'identity display' in NOTIFY messages.David M. Lee
XML encoding in chan_sip is accomplished by naively building the XML directly from strings. While this usually works, it fails to take into account escaping the reserved characters in XML. This patch adds an 'ast_xml_escape' function, which works similarly to 'ast_uri_encode'. This is used to properly escape the local_display attribute in XML formatted NOTIFY messages. Several things to note: * The Right Thing(TM) to do would probably be to replace the ast_build_string stuff with building an ast_xml_doc. That's a much bigger change, and out of scope for the original ticket, so I refrained myself. * It is with great sadness that I wrote my own ast_xml_escape function. There's one in libxml2, but it's knee-deep in libxml2-ness, and not easily used to one-off escape a string. * I only escaped the string we know is causing problems (local_display). At least some of the other strings are URI-encoded, which should be XML safe. Rather than figuring out what's safe and escaping what's not, it would be much cleaner to simply build an ast_xml_doc for the messages and let the XML library do the XML escaping. Like I said, that's out of scope. (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by: Guenther Kelleter Review: http://reviewboard.digium.internal/r/365/ ........ Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378934 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-11Retain XMPP filters across reconnections so external modules continue to ↵Joshua Colp
function as expected. Previously if an XMPP client reconnected any filters added by an external module were lost. This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling. (closes issue ASTERISK-20916) Reported by: kuj ........ Merged revisions 378917 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-11Add JSON API for Asterisk.David M. Lee
This provides a JSON API by pulling in and wrapping the Jansson JSON library[1]. The Asterisk API basically mirrors the Jansson functionality, with a few minor tweaks. * Some names have been asteriskified to protect the innocent. * Jansson provides both reference-stealing and reference-borrowing versions of several API's. The Asterisk API is exclusively reference-stealing for operations that put elements into arrays and objects. * No support for doubles, since we usually don't need that. * Coming along for the ride is the ast_test_validate macro, which made the unit tests much easier to write. [1]: http://www.digip.org/jansson/ (issue ASTERISK-20887) (closes issue ASTERISK-20888) Review: https://reviewboard.asterisk.org/r/2264/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-10* Simplify native bridge code in ast_channel_bridge().Richard Mudgett
* Fix an unbalanced manager_bridge_event(unlink) call if AST_SOFTHANGUP_UNBRIDGE is set in ast_channel_bridge(). * Make ast_channel_bridge() use common cleanup code when leaving the bridge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-10* Removed some noop code and restructured an else-if ladder in ↵Richard Mudgett
ast_generic_bridge(). * Trivial changes in ast_channel_bridge(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09* Simple optimization of bridge_playfile().Richard Mudgett
* Squeezed some redundancy out of update_bridge_vars(). * Wrapped long line in __ast_change_name_nolink(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09Trivial misc bridge code changes.Richard Mudgett
* softmix_bridge_thread() was redundantly initializing an 8K buffer. * Promoted a debug message to a warning in multiplexed_add_or_remove(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09Fix logger.c function definition.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09Trivial misc bridge code changes.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09Tweaked __ast_test_suite_assert_notify() and __ast_test_suite_event_notify() ↵Richard Mudgett
to be void functions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09* Whitespace changes.Richard Mudgett
* Made ast_test_init() match its prototype. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09* Found some more places to use ast_channel_lock_both().Richard Mudgett
* Minor optimization in ast_rtp_instance_early_bridge(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09Fix end condition in ast_rtp_lookup_mime_multiple2.David M. Lee
The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag in the debug output. (closes issue ASTERISK-20772) Reported by: Xavier Hienne ........ Merged revisions 378776 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378780 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09Move declaration of ast_regex_string_to_regex_pattern futher down strings.h.David M. Lee
The prior location is before the declaration of struct ast_str, which causes compiler warnings. (closes issue ASTERISK-20852) Reported by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller (license 6302) ........ Merged revisions 378747 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09Replace errant tabs with spaces in causes.h.David M. Lee
(closes issue ASTERISK-20826) Reported by: snuffy Patches: notabs.dif uploaded by snuffy (license 5024) ........ Merged revisions 378733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378734 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-09app_queue: Fix incorrect assertion.Richard Mudgett
(issue ASTERISK-16115) ........ Merged revisions 378689 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378690 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-08app_queue: Fix multiple calls to a queue member that is in only one queue.Richard Mudgett
When ringinuse=no queue members can receive more than one call if these calls happen at nearly the same time. * Fix so a queue member does not receive more than one call from a queue. NOTE: This fix does not prevent multiple calls to a member if the member is in more than one queue. * Did some refactoring to eliminate some code redundancy. (issue ASTERISK-16115) Reported by: nik600 Patches: jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett Modified * Revert the -r341580 and -r341599 changes adding the queues.conf check_state_unknown option as it was added in an attempt to fix this problem. The fix did not need to be optional. The fix should not have tried to explicitly set the device state. Setting the device state by something other than the device introduces a race condition. I also could not see how the change would be effective other than delaying the app_queue code long enough for the device state to propagate to app_queue. ........ Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378683 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378687 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-06Skinny blob cleanupDamien Wedhorn
Cleanup of red blobs in chan_skinny and possible other small formatting issues. Review: https://reviewboard.asterisk.org/r/2262/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-06Add group and namedgroup pickup to skinnyDamien Wedhorn
Above says it all. Code by snuff, cleaned up by me. Review: https://reviewboard.asterisk.org/r/2246/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-06Rewrite skinny dialing to remove threaded simpleswitchDamien Wedhorn
This rewrite changes skinny dialing from the threaded simpleswitch to a scheduled timeout approach. There were some underlying issues with the threaded simple switch with occasional corruption and possible segfaults. Review: https://reviewboard.asterisk.org/r/2240/ ........ Merged revisions 378622 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04res_srtp: Prevent a crash from occurring due to srtp_create failures in ↵Jonathan Rose
srtp_create Under some circumstances, libsrtp's srtp_create function deallocates memory that it wasn't initially responsible for allocating. Because we weren't initially aware of this behavior, this memory was still used in spite of being unallocated during the course of the srtp_unprotect function. A while back I made a patch which would set this value to NULL, but that exposed a possible condition where we would then try to check a member of the struct which would cause a segfault. In order to address these problems, ast_srtp_unprotect will now set an error value when it ends without a valid SRTP session which will result in the caller of srtp_unprotect observing this error and hanging up the relevant channel instead of trying to keep using the invalid session address. (closes issue ASTERISK-20499) Reported by: Tootai Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header ........ Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378592 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04Fix pjproject compilation in certain circumstancesKinsey Moore
On a fresh checkout of Asterisk 11, running make before ./configure could cause the pjproject subdirectory to get in an odd state that would prevent compilation. This patch by Tilghman prevents that from occurring. (closes issue ASTERISK-20681) Reported by: Dinesh Ramjuttun Tested by: danilo borges, Steve Lang patches: 20121208__ccar_solved.diff.txt uploaded by Tilghman Lesher (license 5003) ........ Merged revisions 378582 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04Fix SIP Notify Messages To Have The Proper IP Address In The FROM FieldMichael L. Young
On a multihomed server when sending a NOTIFY message, we were not figuring out which network should be used to contact the peer. This patch fixes the problem by calling ast_sip_ouraddrfor() and then build_via() so that our NOTIFY message contains the correct IP address. Also, a debug message is being added to help follow the call-id changes that occur. This was helpful for confirming that the IP address was set properly since the call-id contains the IP address. It also will be helpful for troubleshooting purposes when following a call in the debug logs. (closes issue ASTERISK-20805) Reported by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches: asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2255/ ........ Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378559 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04Don't pass STUN packets through the SRTP unprotect function.Joshua Colp
(closes issue AST-1036) Reported by: jbigelow ........ Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378555 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04Doxygen CleanupsAndrew Latham
Baseline clean up of formating to make room for extended documentation (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension PresentMichael L. Young
When the "h" extension is present within the context of the queue, all calls are being reported COMPLETECALLER even when the agent is hanging up the call. This patch checks to see if the agent hung-up or not instead of only relying on checking if the queue (caller) channel hung-up or not. It would appear that having the h extension in the mix, the pbx goes to the h extension, "hanging-up" the queue channel and triggering the reporting of COMPLETECALLER. (closes issue ASTERISK-20743) Reported by: call Tested by: call, Michael L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2256/ ........ Merged revisions 378514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378515 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03chan_agent: Fix wrapup time wait response.Richard Mudgett
* Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup time expires. agent_cont_sleep() had tried but returned the wrong value to stop waiting. * Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void pointer for better type safety. ........ Merged revisions 378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378487 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03Add missing test eventKinsey Moore
This test event was missing from channel.c causing the dial_LS_options test to fail intermittently because of a race condition where most code paths emitted the test event but this one did not. The dial_LS_options test should stop bouncing now. ........ Merged revisions 378455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378459 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03chan_agent: Misc code cleanup.Richard Mudgett
* Fix off-nominal path resource cleanup in agent_request(). * Create agent_pvt_destroy() to eliminate inlined versions in many places. * Pull invariant code out of loop in add_agent(). * Remove redundant module user references in login_exec(). * Remove unused struct agent_pvt logincallerid[] member. ........ Merged revisions 378456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378457 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03chan_agent: Fix agent_indicate() locking.Richard Mudgett
Avoid deadlock potential with local channels and simplify the locking. ........ Merged revisions 378427 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378428 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03Add aliases to the Directory.Tilghman Lesher
This is an interesting feature that allows additional strings to be used to search the Directory, primarily intended to be used with nicknames, but could be used with affiliations and the like. Because the name field is used in more than one place (such as email notifications), it is important that these additional strings not be placed in the name field, but be specified separately. Review: https://reviewboard.asterisk.org/r/2244/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03Prevent exhaustion of system resources through exploitation of event cacheJoshua Colp
This patch changes res_xmpp to no longer cache events under certain circumstances. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore ........ Merged revisions 378411 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03Prevent crashes in res_xmpp when receiving large messagesMatthew Jordan
Similar to r378287, res_xmpp was marshaling data read from an external source onto the stack. For a sufficiently large message, this could cause a stack overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by removing the stack allocation, as it was unnecessary. (issue ASTERISK-20658) Reported by: wdoekes ........ Merged revisions 378409 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02Clean up app_mysql's application entry points to properly parse argumentsMatthew Jordan
When parsing arguments, application entry points should not attempt to directly modify the parameters to the function. This patch properly duplicates the passed in parameters before attempting to parse them. (issue ASTERISK-20658) Reported by: wdoekes patches: issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02Prevent crashes from occurring when reading from data sources with large valuesMatthew Jordan
When reading configuration data from an Asterisk .conf file or when pulling data from an Asterisk RealTime backend, Asterisk was copying the data on the stack for manipulation. Unfortunately, it is possible to read configuration data or realtime data from some data source that provides a large blob of characters. This could potentially cause a crash via a stack overflow. This patch prevents large sets of data from being read from an ARA backend or from an Asterisk conf file. (issue ASTERISK-20658) Reported by: wdoekes Tested by: wdoekes, mmichelson patches: * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674) * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378376 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02Fix AMI redirect action with two channels failing to redirect both channels.Richard Mudgett
The AMI redirect action can fail to redirect two channels that are bridged together. There is a race between the AMI thread redirecting the two channels and the bridge thread noticing that a channel is hungup from the redirects. * Made the bridge wait for both channels to be redirected before exiting. * Made the AMI redirect check that all required headers are present before proceeding with the redirection. * Made the AMI redirect require that any supplied ExtraChannel exist before proceeding. Previously the code fell back to a single channel redirect operation. (closes issue ASTERISK-18975) Reported by: Ben Klang (closes issue ASTERISK-19948) Reported by: Brent Dalgleish Patches: jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/ ........ Merged revisions 378356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378358 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02Prevent exhaustion of system resources through exploitation of event cacheMatthew Jordan
Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) ........ Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378321 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02Resolve crashes due to large stack allocations when using TCPMatthew Jordan
Asterisk had several places where messages received over various network transports may be copied in a single stack allocation. In the case of TCP, since multiple packets in a stream may be concatenated together, this can lead to large allocations that overflow the stack. This patch modifies those portions of Asterisk using TCP to either favor heap allocations or use an upper bound to ensure that the stack will not overflow: * For SIP, the allocation now has an upper limit * For HTTP, the allocation is now a heap allocation instead of a stack allocation * For XMPP (in res_jabber), the allocation has been eliminated since it was unnecesary. Note that the HTTP portion of this issue was independently found by Brandon Edwards of Exodus Intelligence. (issue ASTERISK-20658) Reported by: wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches: ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378287 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-01Add UUID packages now required to configureAndrew Latham
In ASTERISK-20726 UUID was added to Asterisk. This commit is to add the dependancies to the install script git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-01Revert 378248. I changed the logic of this function unitentionally, pointed ↵Sean Bright
out by file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-01Bail out early when building an ast_trans_pvt and the translator doesn't ↵Sean Bright
supply a 'newpvt' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-07 10:03:10 +0000