Before, using the U flag in Voicemail with multiple recipients would put urgent messages
in the INBOX folder for all users past the first thanks to a bug with the message
copying function. This would also cause messages to fail to be sent if the INBOX
directory hadn't been created for that mailbox yet.
(closes issue ASTERISK-18245)
Reported by: Matt Jordan
(closes issue ASTERISK-18246)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1589/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Change from using send() to ast_agi_send() so the HANGUP shows up in the
AGI debug output.
(closes issue ASTERISK-18723)
Reported by: James Van Vleet
Patches:
jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is fortunate that the typo does not alter generated code since the
e->restart.channel and e->ring.channel members are in the same position.
(closes issue ASTERISK-18868)
Reported by: zvision
Patches:
sig_pri.c.diff (License #5755) patch uploaded by zvision
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.
See ASTERISK-18702 it has a very good description of the issue.
I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.
* Added 'dtmf' enum value to sip.conf allowoverlap config option. The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.
* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.
* Fixed get_destination() inconsistency with the pickup extension
matching.
* Fixed initialization of PAGE3 of global_flags in reload_config().
(closes issue ASTERISK-18702)
Reported by: Pavel Troller
Review: https://reviewboard.asterisk.org/r/1517/
Review: https://reviewboard.asterisk.org/r/1582/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
int blah = 1;
...
write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) !=
(new_frames * sizeof(blah)))
is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.
Review: https://reviewboard.asterisk.org/r/1583/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The issue mentioned in the bug report had been fixed recently by
twilson. The reporter included this documentation fix.
(closes issue ASTERISK-18572)
Reported by: Richard Miller
Patch by: Richard Miller (modified)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten
was not expected to be null. This change handles that situation to avoid
a segfault.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change was made because forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the only way voicemail
currently observes whether a mailbox is new or not is by checking to see if the password
is the same as the mailbox number or not.
(closes issue ASTERISK-18282)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1581/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Realtime MOH class caching was implemented because without it, you would build
a completely new MOH class and would start the music over at the beginning each
time hold was pressed in a conversation. Unfortunately, this broke re-scanning
for file changes for realtime MOH classes. This patch corrects that issue.
(closes issue ASTERISK-18039)
Review: https://reviewboard.asterisk.org/r/1579/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes the format type check in ast_closestream and
filestream_destructor. Previously a comparison operator was used, but since
audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes
formats that have a value greater than the video formats), a bitwise AND
operation is used instead. Duplicated code was also moved to filestream_close.
(closes issue ASTERISK-18682)
Reported by: Aldo Bedrij
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1580/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If capability is adjusted when switching to UDPTL during fax transmission, fax
teardown fails. Make sure capability is only touched if RTP is active. This
regression was introduced in R344385.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Holding the channel lock while the CLI "core show channel" command is
executing can slow down the system. It could block the system if the
console output is halted or paused.
* Made capture the CLI "core show channel" output into a buffer to be
output after the channel is unlocked.
* Removed use of C++ keyword as a variable name. out renamed to obuf.
* Checked allocation of obuf for failure so will not crash.
(closes issue ASTERISK-18571)
Reported by: Pavel Troller
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Attempting to call an extension which used Caller ID matching with a channel that
has an empty caller id string would result in a segmentation fault.
(closes issue ASTERISK-18392
Reported By: Ales Zelenik
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks
held. Chan_local attempts to do deadlock avoidance in its ast_call()
callback and could deadlock if a channel lock is already held.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It was strange that the AgentCalled AMI event would get most of its
information from the incoming channel but then get the CallerID
information from the outgoing channel. Before connected line support was
added, this information was always the same at this point.
(closes issue ASTERISK-18152)
Reported by: Thomas Farnham
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes an issue where a user of a dynamic conference was asked for a PIN
twice. This also adds documentation to assist in future modifications to the
piece of code responsible for PIN checking.
(closes issue AST-670)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix bug ASTERISK-16558 which dealt with the order of responses to incoming
streams defined by SDP.
Fix unreported bug where offering multiple same-type streams would cause
Asterisk to reply with an incorrect SDP response missing one or more streams
without a proper declination.
Fix bugs related to a single non-audio stream being offered with responses
requesting codecs that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.
Review: https://reviewboard.asterisk.org/r/1516/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Another deadlock between the conlock/hints and channels/channel locking
orders.
* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().
(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The domain matching code prior to 1.8 used to manually remove the port
from the host:port string when determining if an incoming request
matched the list of domains. When switching to the new parsing
functions, the documentation implied that the "domain" was being
returned by these functions, when instead it was returning the
"hostport" as defined by RFC 3261. This led to confusion and resulted
in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when
domain=x.x.x.x was set in sip.conf.
This patch renames the "domain" variables in the parsing functions to
"hostport" to more accurately describe what it is that they are
returning and also properly truncates the resulting hostport strings
when dealing with domain matching.
Review: https://reviewboard.asterisk.org/r/1574/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The last time this code was touched (by me), a subtlety was missed based on the
difference between needing to check a pin's validity and the need to prompt
for a pin.
(closes issue ASTERISK-18488)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
"dialplan remove include x from y" crashed when the amount of arguments
was less than 6.
(closes issue ASTERISK-18762)
Reported by: Andrey Solovyev
Tested by: Andrey Solovyev
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changing an object value used as a container key requires removing the
object from the container and reinserting it.
* Created change_callid_pvt() to call instead of build_callid_pvt(). The
change_callid_pvt() will correctly change the dialog callid so the ao2
conainter can explicitly unlink it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks.
This function now requires that both the peer and associated pvt be unlocked
before it is called for cases where peer and peer->mwipvt form a circular
reference.
(closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
because of a deadlock between the dialogs container lock and the RWLOCK of
the events subscription list.
* Create dialogs_to_destroy container to hold dialogs that will be
destroyed.
* Ensure that the event subscription callback will never happen with an
invalid peer pointer by making the event callback removal the first thing
in the peer destructor callback.
(closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Review: https://reviewboard.asterisk.org/r/1564/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(these memleaks stop development of gk codes, now i can continue)
Fix printHandler 'Unbalanced Structure' issues with locking printHandler
data for single thread.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original REGISTERTRYING flag, in addition to being impossible to
check, also encroached on the space for the flag above it. This
patch moves the flags that were below REGISTERTRYING back to where
they were as though we had just removed the REGISTERTRYING option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This option is not only useless, but has been broken since inception since
the flag was never copied from the peer where it is set to the pvt where
it was checked. RFC 3261 specificially states that you should not send a
provisional response to a non-INVITE request, and if we did fix the code
so that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This patch
removes registertrying option and any code that would have sent a 100
response to a register.
Review: https://reviewboard.asterisk.org/r/1562/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changeset r342927 introduced a warning which was only supposed to be
emitted when a found realtime peer had an empty (or no) name. It turned
out that there were some inconsistencies left. Now found peers with an
empty name are explicitly ignored like before r342927 but better.
Reviewed by: Stefan Schmidts, Terry Wilson
Review: https://reviewboard.asterisk.org/r/1560
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Integers should always be aligned. For some platforms (ARM, SPARC) this
is more important than for others. This changeset ensures that the
string field string lengths are aligned on *all* platforms, not just on
the SPARC for which there was a workaround. It also fixes that the
length integer can be resized to 32 bits without problems if needed.
(closes issue ASTERISK-17310)
Reported by: radael, S Adrian
Reviewed by: Tzafrir Cohen, Terry Wilson
Tested by: S Adrian
Review: https://reviewboard.asterisk.org/r/1549
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In queues.conf.sample the leastrecent strategy was incorrectly described. Now updated
to reflect how the strategy actually checks peers.
(closes issue ASTERISK-17854)
Reported by: Sebastian Denz
Patches:
queues.conf-doc_issue.patch (License #6139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The MeetMe application documentation has some comments about usage of DAHDI,
and they were a bit outdated relative to modern DAHDI releases. This patch
changes the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since that module
does not exist in current DAHDI releases.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were several problems with the dynamic realtime peer/user lookup
code. The lookup logic had become rather hard to read due to lots of
incremental changes to the realtime_peer function. And, during the
addition of the sipregs functionality, several possibilities for memory
leaks had been introduced. The insecure=port matching has always been
broken for anyone using the sipregs family. And, related, the broken
implementation forced those using sipregs to *still* have an ipaddr
column on their sippeers table.
Thanks Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which caused
the realtime_peer to have a completely unused code path.
This changeset fixes the leaks, the lookup inconsistenties and that
you won't need an ipaddr column on your sippeers table anymore (when
you're using sipregs). Beware that when you're using sipregs, peers
with insecure=port will now start matching!
(closes issue ASTERISK-17792)
(closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes
Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1395
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
