[asterisk-commits] bebuild: tag 1.8.21.0-rc1 r380524 - /tags/1.8.21.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 30 11:51:56 CST 2013
Author: bebuild
Date: Wed Jan 30 11:51:52 2013
New Revision: 380524
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=380524
Log:
Importing files for 1.8.21.0-rc1 release.
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tags/1.8.21.0-rc1/ChangeLog (with props)
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--- tags/1.8.21.0-rc1/ChangeLog (added)
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+2013-01-30 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.21.0-rc1 Released.
+
+2013-01-30 17:44 +0000 [r380451-380520] Matthew Jordan <mjordan at digium.com>
+
+ * configure, configure.ac: Support building Asterisk for Raspberry
+ Pi/Raspbian with hard-float support Building Asterisk on Raspbian
+ with hard-float support fails as it uses the string
+ 'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'.
+ This patch modifies the configure script for Asterisk such that
+ it will match on any string beginning with 'linux-gnueabi', as
+ opposed to requiring an explicit match. (closes issue
+ ASTERISK-21006) Reported by: Christian Hesse Tested by: Christian
+ Hesse patches: linux-gnueabihf.patch uploaded by Christian Hesse
+ (license 6459) linux-gnueabihf-autoconf.patch uploaded by
+ Christian Hesse (license 6459)
+
+ * channels/chan_sip.c: Perform case insensitive comparisons for
+ T.38 attributes RFC5347 section 2.5.2 states the following: ...
+ The attribute "T38MaxBitRate" was once incorrectly registered
+ with IANA as "T38maxBitRate" (lower-case "m"). In accordance with
+ T.38 examples and common implementation practice, the form
+ "T38MaxBitRate" SHOULD be generated by implementations conforming
+ to this package. In general, it is RECOMMENDED that
+ implementations of this package accept lowercase, uppercase, and
+ mixed upper/lowercase encodings of all the T.38 attributes. ...
+ Asterisk currently does not perform case insensitive matching on
+ the T.38 attributes. This causes the T38MaxBitRate attribute to
+ be negotiated at 2400 baud instead of 14400 (or whatever value
+ you actually wanted). This patch makes it so that when we compare
+ T.38 attributes, we do so in a case insensitive fashion. Note
+ that while the issue reporter did not directly write the patch,
+ they contributed to it (and would have provided one themselves if
+ the license had gone through a tad faster), and hence get
+ attribution for it. (closes issue ASTERISK-20897) Reported by:
+ Eric Hill Tested by: Eric Hill patches: -- uploaded by Eric Hill
+
+ * res/res_calendar_icalendar.c: Fix memory leak in
+ res_calendar_icalendar The ICalendar module had a systemic memory
+ leak on each fetch of data from the ICalendar source. The
+ previous fetched data was not being properly disposed. This patch
+ makes it so that before each fetch of data, we dispose of the
+ previously fetched data. (closes issue ASTERISK-21012) Reported
+ by: Joel Vandal Tested by: Joel Vandal
+
+2013-01-29 17:22 +0000 [r380364] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_agent.c: chan_agent: Prevent multiple channels from
+ logging in as the same agent. Multiple channels logging in as the
+ same agent can result in dead channels waiting for a condition
+ signal that will never come because another channel thread stole
+ it. A symptom is chan_sip repeatedly generating warning messages
+ about rescheduling autodestruction of dialogs with an agent
+ channel owner. * Made only login_exec() (the app AgentLogin)
+ clear the agent_pvt->chan pointer to prevent multiple channels
+ from logging in as the same agent. agent_read(), agent_call(),
+ and agent_set_base_channel() no longer disconnect the agent
+ channel from the agent_pvt. This also eliminates the need to keep
+ checking for agent_pvt->chan being NULL. * Made agent_hangup()
+ not wake up the AgentLogin agent thread until it is done. * Made
+ agent_request() not able to get the agent until he has logged in
+ and any wrapup time has expired. * Made agent_request() use
+ ast_hangup() instead of agent_hangup() to correctly dispose of a
+ channel. * Removed agent_set_base_channel(). Nobody calls it and
+ it is a bad thing in general. * Made only agent_devicestate()
+ determine the current device state of an agent. Note: Agent group
+ device states have never been supported. Review:
+ https://reviewboard.asterisk.org/r/2260/
+
+2013-01-29 17:05 +0000 [r380347] David M. Lee <dlee at digium.com>
+
+ * channels/sip/sdp_crypto.c: Corrected crypto tag in SDP ANSWER for
+ SRTP. (again) The original fix (r380043) for getting Asterisk to
+ respond with the correct tag overlooked some corner cases, and
+ the fact that the same code is in 1.8. This patch moves the
+ building of the crypto line out of sdp_crypto_process(). Instead,
+ it merely copies the accepted tag. The call to sdp_crypto_offer()
+ will build the crypto line in all cases now, using a tag of "1"
+ in the case of sending offers. (closes issue ASTERISK-20849)
+ Reported by: José Luis Millán Review:
+ https://reviewboard.asterisk.org/r/2295/
+
+2013-01-29 02:02 +0000 [r380297] Matthew Jordan <mjordan at digium.com>
+
+ * autoconf/ast_check_pwlib.m4, configure: Update configure script
+ to be compatible with ptlib 2.10.9 With ptlib 2.10.9, the
+ configure script fails due to grep returning multiple matches for
+ the pattern it searches for. This patch updates the pattern
+ matching to return only the actual version for the symbol
+ searched for, PTLIB_VERSION. (closes issue ASTERISK-20980)
+ Reported by: Stefan Reuter patches: ASTERISK-20980-1.patch
+ uploaded by Stefan Reuter (license 5339)
+
+2013-01-28 21:06 +0000 [r380254] Sean Bright <sean at malleable.com>
+
+ * channels/chan_iax2.c, channels/iax2.h: Correct the number of
+ available call numbers in IAX2. There is currently an edge case
+ where call number 32768 might be allocated for a call, even
+ though the IAX2 protocol requires call numbers be only 15 bits.
+ This resulted in some unpredictable behavior when call number
+ 32678 is chosen. This patch was mostly written by Richard Mudgett
+ via ReviewBoard. I'm just committing it. Review:
+ https://reviewboard.asterisk.org/r/2293/
+
+2013-01-28 01:52 +0000 [r380210] Russell Bryant <russell at russellbryant.com>
+
+ * main/file.c: Change cleanup ordering in filestream destructor.
+ This patch came about due to a problem observed where wav files
+ had an empty header. The header is supposed to be updated in
+ wav_close(). It turns out that this was broken when the
+ cache_record_files option from asterisk.conf was enabled. The
+ cleanup code was moving the file to its final destination
+ *before* running the close() method of the file destructor, so
+ the header didn't get updated. Another problem here is that the
+ move was being done before actually closing the FILE *. Finally,
+ the last bug fixed here is that I noticed that wav_close() checks
+ for stream->filename to be non-NULL. In the previous cleanup
+ order, it's checking a pointer to freed memory. This doesn't
+ actually cause anything to break, but it's treading on dangerous
+ waters. Now the free() of stream->filename is happening after the
+ format module's close() method gets called, so it's safer.
+ Review: https://reviewboard.asterisk.org/r/2286/
+
+2013-01-23 00:19 +0000 [r379963] Richard Mudgett <rmudgett at digium.com>
+
+ * main/astobj2.c: Attempt to be more helpful when using a bad ao2
+ object pointer. Backport of -r360626 with some enhancements. Put
+ the external obj pointer in the message instead of the internal
+ version.
+
+2013-01-22 18:21 +0000 [r379885] Jonathan Rose <jrose at digium.com>
+
+ * sounds/Makefile, apps/app_meetme.c: app_meetme: Use new prompts
+ for administrator menu The old prompts for the administrator menu
+ were inadequate. They didn't mention that the menu had additional
+ options through the 8 key and pressing the 8 key wouldn't reveal
+ what those options were. This patch fixes all of that while also
+ organizing code pertaining to each individual menu type which was
+ previously all stored in one gigantic function along with many of
+ the basic conference functions. (closes issue AST-996) Reported
+ by: John Bigelow Review:
+ http://reviewboard.digium.internal/r/360/
+
+2013-01-22 14:43 +0000 [r379760-379825] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_meetme.c: Fix station ringback; trunk hangup issues in
+ SLA This patch fixes two bugs: * If an outbound call is made from
+ a SLA phone using SLAStation, then there is no ringtone audible
+ to the phone that originates the call. The indication of the
+ ringing was not being passed to the SLA station; this patch fixes
+ that by passing through the progress indications. * If an SLA
+ station hangs up before the called party answers, then the
+ channel to the called party continues to ring until a timeout
+ occurs. If the called party manages to answer, Asterisk attempts
+ to connect the called party to a non-existant MeetMe room. This
+ patch corrects the behavior by abandoning the call attempt if it
+ detects that the SLA station is no longer in use while attempting
+ to call the called party. Review:
+ https://reviewboard.asterisk.org/r/2275/ (closes issue
+ ASTERISK-20462) Reported by: dkerr patches:
+ asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
+ 5558) asterisk-11-bugid20462.patch uploaded by dkerr (license
+ 5558) (closes issue ASTERISK-20440) Reported by: dkerr patches:
+ asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
+ asterisk-11-bugid20440+20462.patch uploaded by dkerr (license
+ 5558)
+
+ * UPGRADE.txt, contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.slackware.asterisk,
+ contrib/init.d/rc.archlinux.asterisk,
+ contrib/scripts/safe_asterisk, main/asterisk.c,
+ contrib/init.d/rc.suse.asterisk,
+ contrib/init.d/rc.mandriva.asterisk,
+ contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.redhat.asterisk: Update init.d scripts to
+ handle stderr; readd splash screen for remote consoles When
+ r376428 was commited to re-order start up sequences to be more
+ tolerant of forking with thread primitives, a few items were
+ changed that caused changes in behavior on some distros. This
+ includes: * Not displaying the splash screen on a remote console.
+ * Displaying an error message on stderr when a remote console
+ cannot connect to a running instance of Asterisk. In the first
+ case, the splash screen was re-added (thanks to Michael L.
+ Young). In the second case, the various init.d scripts were
+ modified to pipe stderr to /dev/null, as the error message is
+ useful - if you execute a remote console or a remote console
+ command execution and it fail, it should tell you. Note that the
+ error message was always present, it just failed to be printed
+ prior to r376428. Much thanks to the folks who quickly reported
+ this problem, provided solutions, and promptly tested the various
+ init.d scripts on a variety of distros. (closes issue
+ ASTERISK-20945) Reported by: Warren Selby Tested by: Michael L.
+ Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan patches:
+ asterisk-20945-remote-intro-msg.diff uploaded by elguero (license
+ 5026) ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan
+ (license 6283)
+
+2013-01-21 18:27 +0000 [r379718] Kinsey Moore <kmoore at digium.com>
+
+ * codecs/codec_ilbc.c: Prevent segfault for interpolated iLBC
+ frames When iLBC is being used with a jitter buffer and the jb
+ has to interpolate frames, it generates frames with a null
+ pointer and a non-zero datalen. This is now handled properly.
+ (closes issue ASTERISK-20914) Reported By: John McEleney Patches:
+ ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)
+
+2013-01-21 04:59 +0000 [r379645] Andrew Latham <lathama at gmail.com>
+
+ * contrib/scripts/install_prereq: Add LDAP libraries to install
+ script Add LDAP dev package to Debian/Ubuntu install list.
+ Existed in Redhat already. Merged from 11 to Trunk in 379643.
+ Sorry for forgeting 1.8 (issue ASTERISK-20886)
+
+2013-01-21 04:05 +0000 [r379608] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_minivm.c: Fix crash in app_minivm when mime encoding
+ string An incorrect string initializations was left in
+ ast_str_encode_mime from the patch that converted string
+ manipulations to use ast_str strings (r191140). The string
+ initialization causes a crash when ast_str_set is called on the
+ string later on in the function. (closes issue ASTERISK-18697)
+ Reported by: Chris Boot patches:
+ minivm-null-pointer-dereference-fix.patch uploaded by bootc
+ (license 6309) (issue ASTERISK-20854) Reported by: Chris Warr
+ Tested by: Chris Warr
+
+2013-01-19 20:41 +0000 [r379547] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac: Add
+ builtin roundf() for systems lacking it. (closes issue
+ ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276
+ Reported-by: Ovidiu Sas
+
+2013-01-18 23:26 +0000 [r379509] Matthew Jordan <mjordan at digium.com>
+
+ * main/asterisk.c: Fix astcanary startup problem due to wrong pid
+ value from before daemon call When Asterisk forks itself into the
+ background via a call to daemon, it must re-set the pid value of
+ the new process. Otherwise, astcanary gets the pid value of the
+ process before the fork, which prevents it from running. Asterisk
+ eventually starts lowering its priority, as it can no longer
+ communicate with the proverbial canary in the coal mine. This
+ patch ensures that the correct process identifier is used by
+ astcanary. (closes issue ASTERISK-20947) Reported by: Jakob
+ Hirsch Tested by: mjordan patches:
+ asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch
+ (license 6113)
+
+2013-01-18 05:23 +0000 [r379392] David M. Lee <dlee at digium.com>
+
+ * channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
+ channels/sip/reqresp_parser.c: Fix Record-Route parsing for large
+ headers. Record-Route parsing copied the header into a char[256]
+ array, which can be a problem if the header is longer than that.
+ This patch parses the header in place, without the copy, avoiding
+ the issue. In addition to the original patch, I added a unit test
+ for the new get_in_brackets_const function. (closes issue
+ ASTERISK-20837) Reported by: Corey Farrell Patches:
+ chan_sip-build_route-optimized-rev1.patch uploaded by Corey
+ Farrell (license 5909) (with minor changes by dlee)
+
+2013-01-17 02:28 +0000 [r379342] Matthew Jordan <mjordan at digium.com>
+
+ * addons/chan_mobile.c: Fix issue where chan_mobile fails to bind
+ to first available port Per the bluez API, in order to bind to
+ the first available port, the rc_channel field of the socket
+ addressing structure used to bind the socket should be set to 0.
+ Previously, Asterisk had set the rc_channel field set to 1,
+ causing it to connect to whatever happens to be on port 1. We
+ could probably not explicitly set rc_channel to 0 since we memset
+ the struct earlier, but explicitly setting it will hopefully
+ prevent someone from coming in and setting it to some explicit
+ port in the future. (closes issue ASTERISK-16357) Reported by:
+ challado Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin,
+ eliafino, David van Geyn patches: ASTERISK-16357.diff uploaded by
+ Nikolay Ilduganov (license 6253)
+
+2013-01-16 22:45 +0000 [r379310] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: Further fix misinformation in the description of
+ manager MailboxStatus command. The description still claimed that
+ it returned the number of messages rather than whether there were
+ messages waiting.
+
+2013-01-16 21:12 +0000 [r379276] Jason Parker <jparker at digium.com>
+
+ * contrib/scripts/install_prereq: Reduce number of packages
+ install_prereq installs on Debian systems. 'search' will look for
+ any package containing the name provided, so we need to force a
+ more exact search.
+
+2013-01-16 17:40 +0000 [r379226] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: chan_misdn: Fix compile error. (issue
+ ASTERISK-15456)
+
+2013-01-16 04:10 +0000 [r379091-379178] Matthew Jordan <mjordan at digium.com>
+
+ * addons/chan_mobile.c: Fix parsing SMSSRC for SMS messages 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)
+
+ * channels/chan_misdn.c: Set the INVALID_EXTEN channel variable
+ when chan_misdn forces the 'i' extension 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)
+
+ * bridges/bridge_softmix.c: Prevent crash in ConfBridge due to race
+ condition when channels leave bridge When a channel leaves a
+ bridge, a race condition existed where the bridge_channel's pvt
+ structure would be accessed after it was disposed of. This patch
+ prevents that by setting the pointer to the pvt to NULL prior to
+ disposing of it. Note that this patch is a backport from Asterisk
+ 10. This particular race condition was fixed as part of the
+ larger code rework that occurred for that release. The solution
+ to this problem was pointed out by Gunnar Harms in
+ ASTERISK-16640. (closes issue ASTERISK-16640) Reported by:
+ thomas987 (closes issue ASTERISK-16835) Reported by: saghul
+
+2013-01-14 15:11 +0000 [r379001] David M. Lee <dlee at digium.com>
+
+ * channels/chan_sip.c: Fix XML encoding of 'identity display' in
+ NOTIFY messages, continued. 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
+
+2013-01-13 21:15 +0000 [r378967] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c: Reset RTP timestamp; sequence number on
+ SSRC change 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)
+
+2013-01-12 06:26 +0000 [r378933] David M. Lee <dlee at digium.com>
+
+ * main/utils.c, include/asterisk/utils.h, channels/chan_sip.c,
+ tests/test_xml_escape.c (added): Fix XML encoding of 'identity
+ display' in NOTIFY messages. 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
+
+2013-01-09 20:26 +0000 [r378733-378776] David M. Lee <dlee at digium.com>
+
+ * main/rtp_engine.c: Fix end condition in
+ ast_rtp_lookup_mime_multiple2. 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
+
+ * include/asterisk/causes.h: Replace errant tabs with spaces in
+ causes.h. (closes issue ASTERISK-20826) Reported by: snuffy
+ Patches: notabs.dif uploaded by snuffy (license 5024)
+
+2013-01-08 20:22 +0000 [r378663] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c: app_queue: Fix multiple calls to a queue member
+ that is in only one queue. 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
+
+2013-01-04 22:54 +0000 [r378591] Jonathan Rose <jrose at digium.com>
+
+ * res/res_srtp.c: res_srtp: Prevent a crash from occurring due to
+ srtp_create failures in 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
+
+2013-01-04 21:12 +0000 [r378554] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_sip.c: Fix SIP Notify Messages To Have The Proper
+ IP Address In The FROM Field 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/
+
+2013-01-04 21:12 +0000 [r378553] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c: Don't pass STUN packets through the SRTP
+ unprotect function. (closes issue AST-1036) Reported by: jbigelow
+
+2013-01-03 22:09 +0000 [r378514] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/app_queue.c: Fix Queue Log Reporting Every Call
+ COMPLETECALLER With "h" Extension Present 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/
+
+2013-01-03 19:40 +0000 [r378456-378486] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_agent.c: chan_agent: Fix wrapup time wait response.
+ * 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.
+
+ * channels/chan_agent.c: chan_agent: Misc code cleanup. * 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. * Remove some redundant
+ code in agent_request().
+
+2013-01-03 18:35 +0000 [r378455] Kinsey Moore <kmoore at digium.com>
+
+ * main/channel.c: Add missing test event 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.
+
+2013-01-03 17:41 +0000 [r378427] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_agent.c: chan_agent: Fix agent_indicate() locking.
+ Avoid deadlock potential with local channels and simplify the
+ locking.
+
+2013-01-02 21:48 +0000 [r378375] Matthew Jordan <mjordan at digium.com>
+
+ * main/config.c, funcs/func_realtime.c: Prevent crashes from
+ occurring when reading from data sources with large values 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)
+
+2013-01-02 21:08 +0000 [r378356] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h, main/manager.c, main/features.c: Fix
+ AMI redirect action with two channels failing to redirect both
+ channels. 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/
+
+2013-01-02 16:54 +0000 [r378269-378303] Matthew Jordan <mjordan at digium.com>
+
+ * main/devicestate.c, include/asterisk/channel.h,
+ channels/chan_iax2.c, res/res_jabber.c, main/channel.c,
+ channels/chan_dahdi.c, include/asterisk/event_defs.h,
+ channels/chan_skinny.c, main/features.c, main/event.c,
+ apps/app_confbridge.c, funcs/func_devstate.c, res/res_calendar.c,
+ include/asterisk/devicestate.h, channels/chan_local.c,
+ apps/app_meetme.c, channels/chan_sip.c, channels/chan_agent.c:
+ Prevent exhaustion of system resources through exploitation of
+ event cache 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)
+
+ * res/res_jabber.c, channels/sip/include/sip.h,
+ channels/chan_sip.c, main/http.c: Resolve crashes due to large
+ stack allocations when using TCP 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)
+
+2012-12-31 14:41 +0000 [r378217] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Ensure chan_sip rejects encrypted streams
+ without crypto info This ensures that Asterisk rejects encrypted
+ media streams (RTP/SAVP audio and video) that are missing
+ cryptographic keys and ensures that the incoming SDP is
+ consistent with RFC4568 as far as having a crypto attribute
+ present for any SAVP streams. Review:
+ https://reviewboard.asterisk.org/r/2204/
+
+2012-12-20 21:38 +0000 [r378164] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c: Give the causes[] a struct name.
+
+2012-12-20 20:26 +0000 [r378147] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/rtp_engine.h: Adjust RTP instance's
+ available_formats callback to return the correct type. The RTP
+ engine public function that gets the available formats expects a
+ format_t to be returned; however when calling into an RTP
+ instance's callback to get the available formats, the callback
+ returned an int. This never was noticed in Asterisk because the
+ two RTP engines included do not provide an available_formats
+ callback. This introduces an API change, and the proposal for
+ this change was brought up on the Asterisk developers mailing
+ list [1]. There was no public objection to this change, so it is
+ now being put in. (closes AST-1054) reported by Doug Bailey [1]
+ http://lists.digium.com/pipermail/asterisk-dev/2012-December/058058.html
+
+2012-12-18 17:35 +0000 [r378119] Kinsey Moore <kmoore at digium.com>
+
+ * main/channel.c: Add test events for time limit-related hangups
+ This patch adds hangup-related test events in order to support
+ testing of time-limited bridges. This aids in testing the S() and
+ L() bridge options. (issue SWP-4713)
+
+2012-12-17 23:07 +0000 [r378088-378092] Richard Mudgett <rmudgett at digium.com>
+
+ * main/loader.c: Fix potential double free when unloading a module.
+
+ * channels/chan_local.c: Make chan_local module references tied to
+ local_pvt lifetime. The chan_local module references were
+ manually tied to the existence of the ;1 and ;2 channel links. *
+ Made chan_local module references tied to the existence of the
+ local_pvt structure as well as automatically take care of the
+ module references. * Tweaked the wording of the local_fixup()
+ failure warning message to make sense. Review:
+ https://reviewboard.asterisk.org/r/2181/
+
+2012-12-14 21:23 +0000 [r378036] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c: app_queue: Revert bad ringinuse=no patch. With
+ the option ringinuse=no set, the patch committed for
+ ASTERISK-16115 causes non-SIP queue members to never be called
+ because the device state is checked after a channel is created to
+ determine if the member is busy. These queue members always get
+ the "Member %s is busy, cannot dial" message. Most channel
+ drivers other than chan_sip use the default device state
+ handling. The default device-state state is considered in use or
+ unknown if the channel exists or not respectively. (closes issue
+ ASTERISK-20801) Reported by: rmudgett Patches:
+ jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621)
+ patch uploaded by rmudgett
+
+2012-12-13 13:43 +0000 [r377946] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Ensure Min-SE is included in outbound
+ INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
+ value is not 90 (the default) and session timers are not
+ disabled. This has the effect of Asterisk following RFC4028 more
+ closely with regard to 422 responses and preventing situations in
+ which Asterisk would be forced to temporarily accept a call to
+ tear it down based on a Session-Expires below the locally
+ configured Min-SE. (issue SWP-5051) Review:
+ https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
+ Moore Patch-by: Kinsey Moore
+
+2012-12-12 22:39 +0000 [r377922] Rusty Newton <rnewton at digium.com>
+
+ * sounds/Makefile: Incremented EXTRA_SOUNDS_VERSION in
+ sounds/Makefile to 1.4.12 for new Extra Sounds releases See
+ CHANGES-* files in English extra 1.4.12 tarballs for new sound
+ prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan
+ (closes AST-755) Reported by: John Bigelow
+
+2012-12-11 21:54 +0000 [r377847-377881] Richard Mudgett <rmudgett at digium.com>
+
+ * main/aoc.c, main/image.c, main/cel.c, main/timing.c,
+ main/channel.c, main/data.c, main/stun.c, main/file.c,
+ main/http.c: Cleanup CLI commands on exit for several files.
+ (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+ unregister-cli-multiple-all.patch (license #5909) patch uploaded
+ by Corey Farrell
+
+ * main/udptl.c: Cleanup udptl on exit. * Cleanup CLI commands on
+ exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+ udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
+ Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified
+
+2012-12-11 20:45 +0000 [r377840] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_clialiases.c: Fix crash that can occur if CLI
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