[asterisk-dev] ChangeLog format
Paul Belanger
paul.belanger at polybeacon.com
Thu Sep 13 20:26:33 CDT 2012
I'm sure there are other people, besides me, that think are current
ChangeLog format needs a little help. Lets use
ChangeLog-1.8.17.0-rc1[1] for example. All of our comment messages are
smashed together into big blobs of data. Not very helpful.
Compared it to the attached (ChangeLog-1.8.17.0-rc1.svn2cl.txt) version
which I ran through svn2cl[2]; cleaner IMO. Aside from some formatting
with the commit author (accepts an author file) I'm pretty happy how it
looks.
Thoughts? Any other information you'd like to see, or remove?
PS. I snipped the file size to 40k
[1]
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.17.0-rc1
[2] http://arthurdejong.org/svn2cl/
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
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2012-09-12 15:42 mjordan
* [r372959] Constify __ao2_ref_debug in astobj2
When REF_DEBUG is enabled in certain files - most notably ccss.c - the 'tag'
parameter passed to __ao2_ref_debug will be a const char *. The function
currently expects that parameter to not be const. This causes a warning
when compiling, as the const qualifier is being discarded. With dev-mode
enabled, this prevents compiling Asterisk.
This patch makes __ao2_ref_debug's tag and file parameters const.
(closes issue ASTERISK-20408)
Reported by: mjordan
2012-09-12 14:51 mmichelson
* [r372932] Add channel name to a warning to make debugging easier.
The "autodestruct with owner in place" message is typically
indicative of a channel reference leak. Printing out the name
of the channel in the message may be helpful when trying to
debug the issue.
2012-09-11 22:11 jrose
* [r372902] chan_local: Switch from using a random 4 digit hex identifier to
unique id
Changes chan_local channels to use an 8 digit hex identifier generated
atomically and sequentially in order to eliminate the chance of having
multiple channels with the same name during high call volume situations.
(issue ASTERISK-20318)
Reported by: Dan Cropp
Review: https://reviewboard.asterisk.org/r/2104/
2012-09-11 15:26 mmichelson
* [r372840] Fix bad channel application data reference.
When channels get bridged due to an AMI bridge action
or a DTMF attended transfer, the two channels that
get bridged have their application data pointing to
the other channel's name. This means that if one channel
is hung up but the other moves on, it means that the
channel that moves on will have its application data
pointing at freed memory.
(issue ASTERISK-20335)
Reported by: aragon
2012-09-10 20:53 kmoore
* [r372804] Ensure iax2 debug output is displayed when expected
When IAX2 debug was changed from iax_showframe to iax_outputframe,
some instances were missed (or added afterward). This was causing
debug output to not be displayed when expected.
(closes issue ASTERISK-20338)
Reported-by: John Covert
Patch-by: John Covert
2012-09-10 18:35 jrose
* [r372765] app_meetme: Document that 'p' option will continue in dialplan.
(closes issue AST-991)
Reported by John Bigelow
2012-09-10 18:31 kmoore
* [r372763] Warn on CLI when UDPTL init fails
This adds a CLI warning when a SDP offer is rejected due to UDPTL
initialization failure. Previously, there was no indication of the
reason for offer rejection in this case.
(closes issue ASTERISK-20357)
Reported-by: Francesco Usseglio Gaudi
2012-09-10 17:07 jrose
* [r372736] Masquerade: Retain parkinglot settings made by CHANNEL function.
Prior to this patch, the user would have a parkinglot set on a channel that
was parked and when the channel was retrieved, any attempt by that channel
to park would simply use the default. This patch makes parkinglot values
set in this way be retained through the masquerade.
(closes issue AST-990)
Reported by: Nick Huskinson
Patches:
masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182)
2012-09-09 01:19 mjordan
* [r372709] Only re-create an SRTP session when needed; respond with correct
crypto policy
In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an
SDP offer and the ability to re-create an SRTP session when the crypto keys
changed. In certain circumstances - most notably when a phone is put on
hold after having been bridged for a significant amount of time - the act
of re-creating the SRTP session causes problems for certain models of phones.
The patch committed in r356604 always re-created the SRTP session regardless
of whether or not the cryptographic keys changed. Since this is technically
not necessary, this patch modifies the behavior to only re-create the SRTP
session if Asterisk detects that the remote key has changed. This allows
models of phones that do not handle the SRTP session changing to continue
to work, while also providing the behavior needed for those phones that do
re-negotiate cryptographic keys.
In addition, in Asterisk 1.8 only, it was found that phones that offer
AES_CM_128_HMAC_SHA1_32 will end up with no audio if the phone is the
initiator of the call. The phone will send an INVITE request specifying
that AES_CM_128_HMAC_SHA1_32 be used for the cryptographic policy; Asterisk
will set its policy to that value. Unfortunately, when the call is Answered
and a 200 OK is sent back to the UA, the policy sent in the response's SDP
will be the hard coded value AES_CM_128_HMAC_SHA1_80. This potentially
results in Asterisk using the INVITE request's policy of
AES_CM_128_HMAC_SHA1_32, while the phone uses Asterisk's response of
AES_CM_128_HMAC_SHA1_80. Hilarity ensues as both endpoints think the other
is crazy.
This patch fixes that by caching the policy from the request and responding
with it. Note that this is not a problem in Asterisk 10 and later, as the
ability to configure the policy was added in that version.
(issue ASTERISK-20194)
Reported by: Nicolo Mazzon
Tested by: Nicolo Mazzon
Review: https://reviewboard.asterisk.org/r/2099
2012-09-08 03:54 dlee
* [r372682] Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c.
Without this flag, those files will compile with the system installed
OpenSSL headers (if they exist). This is a real bummer if a different
path was specified using --with-ssl=
(closes issue ASTERISK-20392)
2012-09-07 23:05 rmudgett
* [r372655] Fix MALLOC_DEBUG version of ast_strndup().
(closes issue ASTERISK-20349)
Reported by: Brent Eagles
2012-09-07 22:06 rmudgett
* [r372628] Remove annoying unconditional debug message from INC/DEC functions.
(closes issue AST-1001)
Reported by: Guenther Kelleter
2012-09-07 21:48 rmudgett
* [r372624] Fix exception path typo in app_queue.c try_calling().
(closes issue ASTERISK-20380)
Reported by: Jeremy Pepper
Patches:
fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper
2012-09-07 21:23 rmudgett
* [r372620] Fix VoicemailUserEntry event headers ServerEmail and MailCommand
reported values.
The AMI action VoicemailUsersList VoicemailUserEntry event headers
ServerEmail and MailCommand did not report the global values if they were
not overridden. The VoicemailUserEntry event header ServerEmail was not
populated with the global value if the voicemail user did not override it.
The VoicemailUserEntry event header MailCommand was never populated with a
value.
* Removed unused struct ast_vm_user member mailcmd[].
(closes issue AST-973)
Reported by: John Bigelow
Tested by: rmudgett
2012-09-07 02:24 mjordan
* [r372581] Free ast_str objects when temp file fails to be created in MiniVM
The previous commit (r372554) was from a patch that was written before
r366880, which ensured that ast_str objects allocated in the sendmail
routine were free'd in off nominal paths. This commit frees the
string objects in the off nominal path introduced in r372554.
(issue ASTERISK-17133)
Reported by: Tzafrir Cohen
2012-09-07 02:09 mjordan
* [r372554] Fix file descriptor leak and pointer scope issue in MiniVM when
sending mail
When MiniVM sends an e-mail and it has the volgain option set, it will spawn
sox in a separate process to handle the manipulation of the sound file. In
doing so, it creates a temporary file. There are two problems here:
1) The file descriptor returned from mkstemp is leaked
2) The finalfilename character pointer points to a buffer that loses scope
once volgain processing is finished.
Note that in r316265, Russell fixed some gcc warnings by using the return
value of the mkstemp call. A warning was placed in minivm that the file
descriptor was going to be leaked. This patch reverts that change, as it
handles the leak and 'uses' the file descriptor returned from mkstemp.
(closes issue ASTERISK-17133)
Reported by: Tzafrir Cohen
patches:
minivm_18501_demo.diff uploaded by Tzafrir Cohen (license #5035)
2012-09-06 21:38 kmoore
* [r372517] Ensure listed queues are not offered for completion
When using tab-completion for the list of queues on "queue reset stats"
or "queue reload {all|members|parameters|rules}", the tab-completion
listing for further queues erroneously listed queues that had already
been added to the list. The tab-completion listing now only displays
queues that are not already in the list.
(closes issue AST-963)
Reported-by: John Bigelow
2012-09-06 18:54 dsessions
* [r372498] LDAP Realtime Peers Cannot Register
Prior to 1.8, it was not necessary for an explicit "type" to be set for an
asterisk LDAP realtime peer. Now the routine find_peer actually checks the
type field during registration and fails to find the peer if it is not set.
The attached patches make the realtime type equal whatever type is being
searched for if the type is 0 upon return from routine build_peer.
(closes issue ASTERISK-17222)
Reported by: John Covert
Patch by: David Vossel
Tested by: Darren Sessions
Review: https://reviewboard.asterisk.org/r/2095/
2012-09-06 15:52 jrose
* [r372471] chan_sip: Note change in behavior to how directmediapermit/deny ACL
works
r366547 introduced a change to the directmedia ACL for chan_sip which
modified the behavior significantly. Prior to the patch, this option would
bridge peers with directmedia if a peer's IP address matched its own
directmedia ACL. After that patch, the peer would check the bridged peer's
ACL instead. This change has been present since 1.8.14.0. That patched failed
to document the change in Upgrade.txt, so this patch adds mention of that
change to UPGRADE.txt (UPGRADE-1.8.txt in newer branches)
(issue AST-876)
2012-09-06 14:28 kmoore
* [r372444] Ensure "rules" is tab-completable for "queue show"
Previously, tabbing at the end of "queue show" produced a list of
available queues about which information could be shown, but did not
include an alternative command, "rules", to access information about
queue rules. The "rules" item should now be shown in the list of
tab-completable items.
(closes issue AST-958)
Reported-by: John Bigelow
2012-09-06 02:48 mjordan
* [r372417] Fix DUNDi message routing bug when neighboring peer is unreachable
Consider a scenario where DUNDi peer PBX1 has two peers that are its neighbors,
PBX2 and PBX3, and where PBX2 and PBX3 are also neighbors. If the connection
is temporarily broken between PBX1 and PBX3, PBX1 should not include PBX3 in
the list of peers it sends to PBX2 in a DPDISCOVER message, as it cannot send
messages to PBX3. If it does, PBX2 will assume that PBX3 already received the
message and fail to forward the message on to PBX3 itself. This patch fixes
this by only including peers in a DPDISCOVER message that are reachable by the
sending node. This includes all peers with an empty address
(00:00:00:00:00:00) and that are have been reached by a qualify message.
This patch also prevents attempting to qualify a dynamic peer with an empty
address until that peer registers.
(closes issue ASTERISK-19309)
Reported by: Peter Racz
patches:
dundi_routing.patch uploaded by Peter Racz (license 6290)
The patch uploaded by Peter was modified slightly for this commit.
2012-09-06 00:54 mjordan
* [r372390] Allow configured numbers for FollowMe to be greater than 90 characters
When parsing a 'number' defined in followme.conf, FollowMe previously parsed
the number in the configuration file into a buffer with a length of 90
characters. This can artificially limit some parallel dial scenarios. This
patch allows for numbers of any length to be defined in the configuration
file.
Note that Clod Patry originally wrote a patch to fix this problem and received
a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256
characters. Instead, the patch being committed duplicates the string in the
config file on the stack before parsing it for consumption by the application.
(closes issue ASTERISK-16879)
Reported by: Clod Patry
Tested by: mjordan
patches:
followme_no_limit.diff uploaded by Clod Patry (license #5138)
Slightly modified for this commit.
2012-09-05 19:20 kmoore
* [r372354] Correct documentation for ModuleLoad AMI action
The documentation incorrectly listed 'rtp' as a reloadable subsystem
and left out many other reloadable subsystems. It is now also
documented that subsystems may only be reloaded, not loaded or
unloaded.
(closes issue AST-977)
Reported-by: John Bigelow
2012-09-05 18:34 alecdavis
* [r372339] dsp.c: in ast_mf_detect_init incorrectly sets goertzel samples to 160,
should be MF_GSIZE
Related https://reviewboard.asterisk.org/r/2097/
2012-09-05 18:29 kmoore
* [r372337] Ensure counts generated in manager_show_dialplan_helper are correct
When manager_show_dialplan_helper was written, the counter increment
for the total number of contexts was placed with the extensions
increment instead of in the enclosing loop. This function should
now generate correct context counts.
(closes issue AST-970)
Reported-by: John Bigelow
2012-09-05 13:13 mjordan
* [r372268] Fix memory leaks in app_voicemail when using IMAP storage or realtime
config
This patch fixes two memory leaks:
1. When find_user is called with NULL as its first parameter, the voicemail
user returned is allocated on the heap. The inboxcount2 function uses
find_user in such a fashion when counting new messages, and fails to free
the resulting voicemail user object.
2. When populate_defaults is called on a voicemail user, it wipes whatever
flags have been set on the object by copying over the global flags object.
If the VM_ALLOCED flag was ste on the voicemail user prior to doing so,
that flag is removed. This leaks the voicemail user when free_user is later
called.
(closes issue ASTERISK-19155)
Reported by: Filip Jenicek
patches:
asterisk.patch2 uploaded by Filip Jenicek (license 6277)
Patch slightly modified for this commit.
Review: https://reviewboard.asterisk.org/r/2096
2012-09-05 07:35 alecdavis
* [r372239] dsp.c: Fix multiple issues when no-interdigit delay is present, and
fast DTMF 50ms/50ms
Revert DTMF hit/miss detector to original -r349249 method with some changes,
remove unnecessary;
1. reseting of hits=0, when no signal, only need to set it once.
2. incrementing of hits, when the hit is the same as the current hit.
3. setting of lasthit, when it's the same as before.
Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3
& 3 spelling mistakes
(closes issue ASTERISK-19610)
alecdavis (license 585)
Reported by: Jean-Philippe Lord
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2085/
2012-09-05 06:45 alecdavis
* [r372212] dsp.c: optimize goerztzel sample loops, in dtmf_detect, mf_detect and
tone_detect
use a temporary short int when repeatedly used to call goertzel_sample.
alecdavis (license 585)
Reported by: alecdavis
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2093/
2012-09-05 03:45 elguero
* [r372185] Fix Incrementing Sequence Number For Retransmitted DTMF End Packets
In Asterisk 1.4+, a fix was put in place to increment the sequence number for
retransmitted DTMF end packets. With the introduction of the RTP engine API in
1.8, the sequence number was no longer being incremented. This patch fixes this
regression as well as cleans up a few lines that were not doing anything.
(closes issue ASTERISK-20295)
Reported by: Nitesh Bansal
Tested by: Michael L. Young
Patches:
01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license 6418)
asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2083/
2012-09-05 02:16 mjordan
* [r372158] Fix memory leak when CEL is successfully written to PostgreSQL
database
PQClear is not called when the result object of a call to PQExec has a
status of PGRES_COMMAND_OK. Interestingly enough, the off nominal case was
handled properly, so this memory leak only occurred when CEL records were
successfully written.
This patch properly clears the result in the nominal code path.
(closes issue ASTERISK-19991)
Reported by: Etienne Lessard
Tested by: Etienne Lessard
patches:
mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license #6394)
2012-08-30 20:51 mmichelson
* [r372089] Prevent crash on shutdown due to refcount error on queues container.
When app_queue is unloaded, the queues container has its refcount
decremented, potentially to 0. Then the taskprocessor responsible
for handling device state changes is unreferenced. If the
taskprocessor happens to be just about to run its task, then it
will create and destroy an iterator on the queues container.
This can cause the refcount on the queues container to increase to
1 and then back to 0. Going back to 0 a second time results in
double frees.
This failure was seen periodically in the testsuite when Asterisk
would shut down.
2012-08-30 18:28 mmichelson
* [r372048] Help prevent ringing queue members from being rung when ringinuse set
to no.
Queue member status would not always get updated properly when the member
was called, thus resulting in the member getting multiple calls. With this
change, we update the member's status at the time of calling, and we also
check to make sure the member is still available to take the call before
placing an outbound call.
(closes issue ASTERISK-16115)
reported by nik600
Patches:
app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409)
2012-08-30 16:21 mjordan
* [r372015] AST-2012-013: Resolve ACL rules being ignored during calls by some
IAX2 peers
When an IAX2 call is made using the credentials of a peer defined in a dynamic
Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are
not applied to the call attempt. This allows for a remote attacker who is aware
of a peer's credentials to bypass the ACL rules set for that peer.
This patch ensures that the ACLs are applied for all peers, regardless of their
storage mechanism.
(closes issue ASTERISK-20186)
Reported by: Alan Frisch
Tested by: mjordan, Alan Frisch
2012-08-30 16:05 mjordan
* [r371998] AST-2012-012: Resolve AMI User Unauthorized Shell Access through
ExternalIVR
The AMI Originate action can allow a remote user to specify information that can
be used to execute shell commands on the system hosting Asterisk. This can
result in an unwanted escalation of permissions, as the Originate action, which
requires the "originate" class authorization, can be used to perform actions
that would typically require the "system" class authorization. Previous attempts
to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought
to do so by inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched a predefined
set of values, rejecting the command if the user lacked the "system" class
authorization. As noted by IBM X-Force Research, the "ExternalIVR"
application is not listed in the predefined set of values. The solution for
this particular vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class authorization.
Unfortunately, the approach of inspecting fields in the Originate action against
known applications/functions has a significant flaw. The predefined set of
values can be bypassed by creative use of the Originate action or by certain
dialplan configurations, which is beyond the ability of Asterisk to analyze at
run-time. Attempting to work around these scenarios would result in severely
restricting the applications or functions and prevent their usage for legitimate
means. As such, any additional security vulnerabilities, where an
application/function that would normally require the "system" class
authorization can be executed by users with the "originate" class authorization,
will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has
been updated to reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper system
configuration can limit the impact of such scenarios.
(closes issue ASTERISK-20132)
Reported by: Zubair Ashraf of IBM X-Force Research
2012-08-30 12:47 mjordan
* [r371961] Restore CODING-GUIDELINES to doc folder
In r294740, the CODING-GUIDELINES was removed from the doc folder in favor
of the content on the Asterisk wiki. Some folks still look in the doc folder
initially for coding guideline suggestions; as such, this patch adds a
CODING-GUIDELINES file back into the doc folder. The content of the file
merely points to the correct page on the Asterisk wiki where the coding
guidelines currently live.
(closes issue ASTERISK-20279)
Reported by: Andrew Latham
Patches:
CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985)
2012-08-29 20:42 jrose
* [r371919] app_meetme: Adding test events for following activity in MeetMe.
2012-08-29 19:38 rmudgett
* [r371888] Initialize file descriptors for dummy channels to -1.
Dummy channels usually aren't read from, but functions like SHELL and CURL
use autoservice on the channel.
(closes issue ASTERISK-20283)
Reported by: Gareth Palmer
Patches:
svn-371580.patch (license #5169) patch uploaded by Gareth Palmer (modified)
2012-08-29 18:22 rmudgett
* [r371860] Fix hangup cause passthrough regression.
The v1.8 -r369258 change to fix the F and F(x) action logic introduced a
regression in passing the hangup cause from the called channel to the
caller channel.
(closes issue ASTERISK-20287)
Reported by: Konstantin Suvorov
Patches:
app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov
(modified)
Tested by: rmudgett
2012-08-29 16:59 jrose
* [r371824] chan_sip: Send 408 on retransmit timeout instead of 603
(closes issue ASTERISK-20124)
Reported by: Walter Doekes
2012-08-27 21:47 mmichelson
* [r371787] Fix misleading documentation in agents.conf.sample regarding ackcall
usage.
The documentation made it sound as if the DTMF acknowledgment was needed
at the time the agent logs in, rather than when the agent is called. This
is likely a relic from the days when there were multiple ways of logging
in agents.
(closes issue AST-962)
reported by Steve Pitts
2012-08-27 21:24 mmichelson
* [r371782] Fix incorrect documentation of the MailboxStatus manager command.
The "Waiting" field was misdocumented as reporting the number of
messages waiting. In reality, it simply indicated the presence or
absence of waiting messages.
(closes issue AST-975)
reported by John Bigelow
2012-08-27 17:35 mmichelson
* [r371747] Fix incorrectly documented option in queues.conf
sharedlastcall defaults to "no" not "yes"
(closes issue AST-979)
reported by Steve Pitts
2012-08-27 16:40 dlee
* [r371718] Fixes ast_rwlock_timed[rd|wr]lock for BSD and variants.
The original implementations simply wrap pthread functions, which take
absolute time as an argument. The spinlock version for systems without
those functions treated the argument as a delta. This patch fixes the
spinlock version to be consistent with the pthread version.
(closes issue ASTERISK-20240)
Reported by: Egor Gorlin
Patches:
lock.c.patch uploaded by Egor Gorlin (license 6416)
2012-08-27 13:43 kmoore
* [r371690] Implement workaround for BETTER_BACKTRACES crash
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in the
testsuite since several tests run "core show locks" to help with
debugging. This seems to be a fault with libraries on certain operating
systems (notably CentOS 6.2/6.3) running on virtual machines and
utilizing gcc 4.4.6.
(closes issue ASTERISK-20090)
2012-08-26 23:03 alecdavis
* [r371662] mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE
2012-08-21 20:35 mmichelson
* [r371590] Fix misuses of asprintf throughout the code.
This fixes three main issues
* Change asprintf() uses to ast_asprintf() so that it
pairs properly with ast_free() and no longer causes
MALLOC_DEBUG to freak out.
* When ast_asprintf() fails, set the pointer NULL if
it will be referenced later.
* Fix some memory leaks that were spotted while taking
care of the first two points.
(Closes issue ASTERISK-20135)
reported by Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2071
2012-08-20 15:25 kmoore
* [r371544] Ignore recovered zero-length secondary UDPTL packets
In some cases, recovering lost packets using the secondary packet
recovery mechanism with UDPTL/T.38 can result in the recovery of
zero-length packets. These must be ignored or the frame generated from
them can cause segfaults and allocation failures.
(closes issue ASTERISK-19762)
(closes issue ASTERISK-19373)
Reported-by: Benjamin (bulkorok)
Reported-by: Rob Gagnon (rgagnon)
2012-08-17 18:51 mjordan
* [r371469] Fix memory leak in XML documentation
When formatting documentation fields, the XML documentation parser calls
xmldoc_get_formatted. This function allocates a string buffer at the
beginning of its routine. Unfortunately, on certain code paths, it also
calls xmldoc_string_cleanup, which assumes that it will create the string
buffer. The previously allocated string buffer is then leaked by the
xmldoc_string_cleanup routine.
Now: we don't do that.
(closes issue AST-932)
Reported by: Alexander Homig
2012-08-17 15:49 kmoore
* [r371436] Add instrumentation to subsystem reloads
When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr,
extconfig, etc.
(issue PQ-1126)
2012-08-16 22:41 kmoore
* [r371393] Add module reload instrumentation for TEST_FRAMEWORK
This adds AMI events for module reloads when Asterisk is built with
TEST_FRAMEWORK enabled and corrects generation of the module load AMI
event.
(issue PQ-1126)
2012-08-16 22:30 twilson
* [r371392] Handle integer over/under-flow in ast_parse_args
The strtol family of functions will return *_MIN/*_MAX on overflow. To
detect when an overflow has happened, errno must be set to 0 before
calling the function, then checked afterward.
(closes issue ASTERISK-20120)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2073/
2012-08-16 18:57 jrose
* [r371357] chan_sip: Use pvt outgoing_call variable to set Remote-Party-ID Header
Previously the pvt SIP_OUTGOING flag was used instead, which will frequently
flip during reinvites.
(closes issue AST-897)
Reported by: Thomas Arimont
2012-08-16 15:46 jrose
* [r371337] chan_sip: Trigger reinvite if the SDP answer is included in the SIP
ACK
Under certain conditions, a SIP transaction involving directmedia wouldn't
trigger a re-invite because the SDP answer was included in an ACK instead
of in a message that we would have triggered the invite with. This patch
just queues a source change control frame if the dialog is using
directmedia when we find sdp for an ACK.
(closes issue AST-913)
Reported by: Thomas Arimont
2012-08-15 23:10 mmichelson
* [r371306] Fix bug where final queue member would not be removed from memory.
If a static queue had realtime members, then there could be a potential
for those realtime members not to be properly deleted from memory.
If the queue's members were loaded from realtime and then all the
members were deleted from the backend, then the queue would still
think these members existed. The reason was that there was a short-
circuit in code such that if there were no members found in the
backend, then the queue would not be updated to reflect this.
Note that this only affected static queues with realtime members.
Realtime queues with realtime members were unaffected by this issue.
(closes issue ASTERISK-19793)
reported by Marcus Haas
2012-08-15 20:14 kmoore
* [r371270] Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog
destruction
The other instance of this bug was fixed by jcolp/file in r121496. If
we are destroying a dialog only set the MWI dialog pointer on the
related peer to NULL if it is the dialog currently being destroyed.
(closes issue ASTERISK-20119)
Patch-by: Misha Vodsedalek
2012-08-13 20:00 kmoore
* [r371201] Add test instrumentation
This adds test instrumentation for loading and unloading of modules
and for certain actions in MeetMe to be used in the testsuite or any
other consumer of AMI events. These will only be generated when
Asterisk is built with TEST_FRAMEWORK enabled.
(issue PQ-1131)
(issue PQ-1133)
2012-08-13 19:49 mmichelson
* [r371198] Fix problem where incorrect pointer was checked for nullity.
2012-08-10 21:21 mmichelson
* [r371141] Fix a couple of documentation problems in app_queue.c
* The RemoveQueueMember app made mention of options that could
be passed in, but no options are supported. I have removed the
listing of options from the documentation.
* The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible
value that could be set.
(closes issue AST-949)
reported by Steve Pitts
(closes issue AST-954)
reported by Steve Pitts
2012-08-10 16:40 may
* [r371089] remove ALREADYGONE flag on ooh323 call data by ooh323_indicate
(CONGESTION/BUSY) due to call hasn't gone there really.
This indication arrive from asterisk core not h.323 stack
(closes issue ASTERISK-19308)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-19308.patch
2012-08-10 15:10 may
* [r371060] Send re-register packets by GRQ (gatekeeper request) interval
(close issue ASTERISK-20094)
Patches:
ASTERISK-20094-2.patch
2012-08-09 18:58 rmudgett
* [r371012] Use better libss7 detection test and move libpri compile test.
2012-08-09 18:58 may
* [r371011] Fix to resend GRQ/RRQ if RRJ (registration reject) is received
(close issue ASTERISK-20094)
Patches:
ASTERISK-20094.patch
2012-08-09 18:02 may
* [r370988] change opening h323 logfile with append mode instead of overwrite
2012-08-09 17:39 kmoore
* [r370985] Correct documentation for the MeetMe x flag
The documentation for the x flag for MeetMe incorrectly described its
function as closing down the conference when the last marked user left.
It actually causes the users with that flag to leave the conference
when the last marked user exits. The functionality of this flag is not
changing.
2012-08-08 22:40 elguero
* [r370952] Fix Not Unreferencing A Spied Channel
When a channel hangs up while being spied upon and the option to exit the
ChanSpy application when the spied on channel hangs up is set,
ast_autochan_destroy is not being called and therefore a reference to the spied
upon channel is not removed.
The symptom being reported was that when using func_group in the dialplan and
calling "group show channels" at the cli, the spied upon channel was still
being shown while "core show channels" showed that the channel was not up.
This patch calls ast_autochan_destroy when a spied upon channel hangs up and
the option to exit the ChanSpy application is set, removing the reference to
the channel allowing the count for the group that the spied channel was part of
to be decremented.
(closes issue ASTERISK-17515)
Reported by: Arkadiusz Malka
Tested by: Alexandr Gordeev, Michael L. Young
Patches:
asterisk-17515-destroy-autochan.diff
uploaded by Michael L. Young (license 5026)
2012-08-08 20:28 kmoore
* [r370923] Do not define a cause that doesn't actually exist
AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause
information. As such, it should not be defined and translatable as a
cause.
2012-08-08 19:58 rmudgett
* [r370900] Fix the analog dial *0 flash-hook of bridged peer feature.
The flash-hook the bridged peer feature now correctly determines if the
bridged peer is another chan_dahdi channel, that it is an analog channel,
and that it has the correct signaling for an FXO port. It now also
flash-hooks the correct channel.
2012-08-07 19:19 kmoore
* [r370856] Add missing AST_CAUSE_* -> text translations
2012-08-06 15:00 mmichelson
* [r370797] Improve debug message for temporary outbound proxies.
Thanks to Paul Belanger for pointing this out.
2012-08-03 21:43 mmichelson
* [r370771] Seriously? Another compilation error fixed.
Somebody beat me.
2012-08-03 21:39 mmichelson
* [r370770] Remove unused variable.
2012-08-03 21:35 mmichelson
* [r370769] Fix error in the "IPorHost" section of a SIP dialstring.
This is based on the review request posted by Walter Doekes
(referenced lower in the commit message)
The main fix here is to treat the IPorHost portion of the dial
string as a temporary outbound proxy. This ensures requests
get sent to the proper location.
Due to the age of the request, some parts were no longer relevant.
For instance, the request moved outbound proxy parsing code into
a single method. This is done in a previous commit, so it was not
necessary to do again.
Also, the review request fixed some errors with regards to request
routing for CANCEL and ACK requests. This has also been fixed in
more recent commits.
(closes issue ASTERISK-19677)
reported by Walter Doekes
Review https://reviewboard.asterisk.org/r/1859
2012-08-01 02:25 kmoore
* [r370697] Revert alloca changes for utils
These changes were a tad overzealous in the utils directory.
Unfortunately, these don't compile with a "make".
2012-07-31 20:54 mjordan
* [r370666] Schedule pokes of registered SIP peers within a given timespan after
SIP reload
With a large number of SIP peers registered, performing a SIP reload causes a
flood of SIP OPTIONS request packets. These are immediately sent out, and, as
responses come back, can cause peers to be flagged as 'lagged' due to handling
of the many response messages.
This fix prevents this "packet storm" and schedules the pokes for a random
time. That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting.
The committed patch has some very small modifications to the patch schmidts
wrote for the review.
(closes issue ASTERISK-19154)
Reported by: Nicolo Mazzon
patches:
issue19154.patch license #6034 uploaded by schmidts
Review: https://reviewboard.asterisk.org/r/1652
2012-07-31 19:31 kmoore
* [r370642] Clean up and ensure proper usage of alloca()
This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().
(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
2012-07-31 15:26 mmichelson
* [r370618] Help mitigate potential reinvite glare scenarios.
When Asterisk servers are set up back-to-back, and
direct media is to be used betweeen endpoints, it is
fairly common for the two Asterisk servers to send
direct media reinvites to each other simultaneously.
This results in 491s and ACKs being exchanged between
the servers. While the media eventually gets set up
properly, the problem is that there can be a noticeable
delay for the streams to stabilize.
This patch adds a new directmedia option called "outgoing".
With this set, an immediate direct media reinvite will only
be sent if the call direction is outgoing. For incoming
dialogs, an immediate direct media reinvite will not be sent,
but further "reactionary" direct media reinvites may be sent.
For those who are having some deja vu, that's because this
patch was originally committed to trunk since there is a
new configuration option added. After seeing a bug report
about audio being slow to set up on SIP calls, it became
apparent that this patch would be the best solution for
resolving the issue. The patch is unintrusive and will
have no effect unless the option is explicitly enabled.
(closes issue AST-896)
reported by Thomas Arimont
(closes issue ASTERISK-19857)
reported by Matt Jordan
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