Age | Commit message (Collapse) | Author |
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
(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
|
|
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
|
|
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
|
|
* 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
|
|
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
|
|
* 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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
out by file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
supply a 'newpvt'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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/
........
Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378218 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378219 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 378164 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378165 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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)
........
Merged revisions 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378120 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378121 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 378092 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378093 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378094 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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/
........
Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378089 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378090 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Fix local_alloc() unexpected limitation of exten and context length from
a combined length of 80 characters to a normal 80 characters each.
* Made local_alloc() and local_devicestate() parse the same way.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This was causing issues when using DESTDIR, since the path to which the link
pointed is not likely to exist (and not useful to exist) on the target system.
(issue ASTNOW-284)
........
Merged revisions 378073 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* awesome_locking() does not need to thrash the pvt lock as much.
* local_setoption() does not need to check for NULL pvt on cleanup since
it will never be NULL.
* Made ref the pvt before locking for consistency.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
No need to check for an agent twice. Santa does that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
does. In addition, the BRIDGE_FEATURES variable would not apply features to
the proper party based on whether the feature option letter was in caps or
in lowercase (both ways would apply it to the caller). Now uppercase applies
to the caller while lowercase applies to the callee (like with the dial option)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
........
Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378037 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378038 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Fixup the vmexten so if globally set in general section will be honored by
chan_skinny. Also get rid of the 'global_' part of variable name to match
regexten.
(closes issue ASTERISK-20790)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-vm.diff uploaded by snuffy (license 5024)
........
Merged revisions 378010 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue ASTERISK-20788)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-g722.diff uploaded by snuffy (license 5024)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When two users entered a new conference simultaneously, one of the callers
hears MOH. This happened if two unmarked users entered simultaneously and
also if a waitmarked and a marked user entered simultaneously.
* Created a confbridge internal MOH API to eliminate the inlined MOH
handling code. Note that the conference mixing bridge needs to be locked
when actually starting/stopping MOH because there is a small window
between the conference join unsuspend MOH and actually joining the mixing
bridge.
* Created the concept of suspended MOH so it can be interrupted while
conference join announcements to the user and DTMF features can operate.
* Suspend any MOH until the user is about to actually join the mixing
bridge of the conference. This way any pre-join file playback does not
need to worry about MOH.
* Made post-join actions only play deferred entry announcement files.
Changing the user/conference state during that time is not protected or
controlled by the state machine.
(closes issue ASTERISK-20606)
Reported by: Eugenia Belova
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2232/
........
Merged revisions 377992 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377993 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and
correct len of 2 strcmp in skinny_setdebug(). (see opticron's review
on https://reviewboard.asterisk.org/r/2240/)
........
Merged revisions 377991 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
I changed this code earlier to return NULL if it wasn't able to generate a UUID,
whereas the earlier code would always return the ast_str that was passed in.
Switch back to returning the ast_str, only set it to the empty string instead if
UUID generation fails. We still do a validity check later which will catch this
and blow up if necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Review the syntax of the 'skinny debug' command to show more than
just 'show' for options to 'skinny debug' command.
(closes issue ASTERISK-20789)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-debug.diff uploaded by snuffy (license 5024)
........
Merged revisions 377985 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The code was doing a runtime check, anyway. The compile time check isn't
always valid (cross-compiling, packages).
Review: https://reviewboard.asterisk.org/r/2245/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Currently the res_calendar_exchange module uses its own method of generating
UUIDs using ast_random(). Now that we have a UUID API we should use that
instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This puts back in the changes that are designed to work
around a memory leak fix in the CLI code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Introduced in r377846, the configure script was looking for uuid.h instead
of uuid/uuid.h.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|