When Asterisk fails to start a PBX thread for a new channel - for example, when
the maxcalls setting in asterisk.conf is exceeded - we currently send a final
response, and then attempt to send a BYE request to the UA. Since that's all
sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt
such that we don't get stuck sending BYE requests to something that does not
want it.
Note that this patch is a slight modification of the one on ASTERISK-15434.
For clarity, it explicitly calls sipalreadygone with the calls to transmit a
final response.
ASTERISK-21845
ASTERISK-15434 #close
Reported by: Makoto Dei
Tested by: Matt Jordan
patches:
sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since Asterisk won't build without the library, not having it is definitely
an error. Thanks to Kyle Kurz for pointing this out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we receive an SDP as part of an offer/answer for a peer/friend has been
configured to require encryption, and that SDP offer/answer failed to provide
acceptable crypto attributes, we currently issue a WARNING that uses the phrase
"we" and "requested". In this case, both of those terms are ambiguous - the
user will probably think "we" is Asterisk (it most likely isn't) and it may
not be a "request", so much as an SDP that was received in some fashion.
This patch makes the WARNING messages slightly less bad and a bit more
accurate as well.
ASTERISK-23214 #close
Reported by: Rusty Newton
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, SDP offers negotiating SDES-SRTP crypto attributes would
be rejected if those crypto attributes contained either a key lifetime or a
MKI parameter. While from a theoretical point of view this was defensible -
Asterisk does not support key lifetimes or multiple crypto keys - from a
practical point of view, this is quite a problem. A large number of endpoints
offer lifetimes/MKI, which Asterisk can tolerate so long as it doesn't actually
have to support anything more than a single key or refresh the key.
In reality, this is (so far as we've seen) always the case.
This patch is a forward port of Olle's work in the lingon-srtp-key-lifetime-1.8
branch. To quote Olle from ASTERISK-17721, it handles lifetime/MKI parameters
in the following fashion:
> The Lingon branch now handle lifetime and MKI parameters.
>
> We only accept lifetimes up to max for the crypto and higher than 10 hours
> for packetization of 20 ms (50 pps).
>
> We only handle MKI with index 1.
>
> We do not really bother with counting packets and reinviting at end of
> lifetime, so the min of 10 hours kind of takes care of most calls. If there
> are longer ones, we rely on the other side for re-invites.
>
> It's still not perfect, but I personally think this is an improvement. A
> configuration option for minimum lifetime accepted could be added.
When the patch was ported forward, I decided against adding a configuration
option as Olle's handling was more than sufficient for every case I've seen
come through the issue tracker or through interoperability testing. We can
revisit that decision if it proves to be false.
A few small other tweaks were made to the surrounding code to reduce
indentation and provide better type safety for the 'tag' parameter.
Review: https://reviewboard.asterisk.org/r/4419/
Review: https://reviewboard.asterisk.org/r/4418/
ASTERISK-17721 #close
Reported by: Terry Wilson
ASTERISK-17899 #close
Reported by: Dwayne Hubbard
patches:
lingon-srtp-key-lifetime-1.8.diff uploaded by oej (License 5267)
ASTERISK-20233
Reported by: tootai
ASTERISK-22748
Reported by: Alejandro Mejia
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some WebSocket applications, like [chan_respoke][], require a larger
frame size than the default 8k; this patch bumps the default to 16k.
This patch also fixes some problems exacerbated by large frames.
The sanity counter was decremented on every fread attempt in
ws_safe_read(), regardless of whether data was read from the socket or
not. For large frames, this could result in loss of sanity prior to
reading the entire frame. (16k frame / 1448 bytes per segment = 12
segments).
This patch changes the sanity counter so that it only decrements when
fread() doesn't read any bytes. This more closely matches the original
intention of ws_safe_read(), given that the error message is
"Websocket seems unresponsive".
This patch also properly logs EOF conditions, so disconnects are no
longer confused with unresponsive connections.
[chan_respoke]: https://github.com/respoke/chan_respoke
Review: https://reviewboard.asterisk.org/r/4431/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the monitor thread is stopped, its pthread ID is set to a specific value
(AST_PTHREADT_STOP) so that later portions of the code can determine whether
or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit
failed to check for that value, checking instead only for AST_PTHREAD_STOP.
Passing the invalid yet very specific value to pthread_kill causes a crash.
This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that
it doesn't attempt to poke the thread if the thread has already been stopped.
ASTERISK-24800 #close
Reported by: JoshE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When more than one call using the same codec type enters into a softmix bridge
and no audio is present for a channel the bridge optimizes the out frame by
using the same one for all channels with the same codec type. Unfortunately,
when that number (channels with same codec type) dropped to <= 1 the codec
was not dereferenced. At least not until all parties left the bridge. Thus in
the case of G.729 the license was not released. This patch ensures that the
codec is dereferenced immediately when the optimization no longer applies.
ASTERISK-24797 #close
Reported by: Luke Hulsey
Review: https://reviewboard.asterisk.org/r/4429/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using IMAP voicemail with FreePBX, you will often get ERROR messages
complaining about not being able to find a mailbox. This is due to how FreePBX
handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this
a configuration error, as in any other system it would be indicative of
someone misconfiguring their system.
Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch
demotes the message so that system administrators can hopefully reduce some
of the noise in their log files.
Note that in the original patch this was made into a NOTICE, but that's a
too forgiving.
ASTERISK-24790 #close
Reported by: Graham Barnett
patches:
app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Change __ast_module_shutdown_ref to be NULL safe (11+).
* Allow modules that call ast_bucket_scheme_register or ast_codec_register
to be unloaded during graceful shutdown only (13+ only).
ASTERISK-24796 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4428/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When interfacing with Microsoft Exchange, custom headers will be returned as
all lower case. Currently, the IMAP header code will fail to parse the returned
custom headers, as it will be performing a case sensitive comparison. This can
cause playback of messages to fail, as needed information - such as origtime -
will not be present.
This patch updates app_voicemail's header parsing code to perform a case
insensitive lookup for the requested custom headers. Since the headers are
specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be
unique in an IMAP message, this should cause no issues with other systems.
ASTERISK-24787 #close
Reported by: Graham Barnett
patches:
app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some distributions are going to disable SSLv3 at compile time. This option can
be checked using the directive OPENSSL_NO_SSL3_METHOD. This patch updates the
TCP/TLS handling in Asterisk to look for that directive before attempting to
use the SSLv3 specific methods.
ASTERISK-24799 #close
Reported by: Alexander Traud
patches:
no-ssl3-method.patch uploaded by Alexander Traud (License 6520)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added ast_sched_clean_by_callback for cleanup of scheduled events
that have not yet fired.
* Run all pending peercnt_remove_cb and replace_callno events in chan_iax2.
Cleanup of replace_callno events is only run 11, since it no longer
releases any references or allocations in 13+.
ASTERISK-24451 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4425/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Test Event for MIXMONITOR_END - which signals that a MixMonitor has
completed - technically fired before the filestream was closed. If a test
used this to trigger a condition to verify that the file was written, it
could result in a race condition where the file size would not be what the
test expected.
Luckily, no tests were using this (although they should have been). Since the
test event needed to be moved after the point where the MixMonitor autochan has
been destroyed, the test event no longer emits the channel name. Luckily,
nothing needs it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a SIP device that has its registration stored in RealTime unregisters,
the entry for that device is updated with blank values, i.e., "", indicating
that it is no longer registered. Unfortunately, one of those values that is
'blanked' is the device's port. If the column type for the port is not a
string datatype (the recommended type is integer), an ODBC or database error
will be thrown. MariaDB does not coerce empty strings to a valid integer value.
This patch updates the query run from chan_sip such that it replaces the port
value with a value of '0', as opposed to a blank value. This is the value that
other database backends coerce the empty string ("") to already, and the
handling of reading a RealTime registration value from a backend already
anticipates receiving a port of '0' from the backends.
ASTERISK-24772 #close
Reported by: Richard Miller
patches:
chan_sip.diff uploaded by Richard Miller (License 5685)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add ast_module_shutdown_ref for use by modules that can
only be unloaded during graceful shutdown.
When REF_DEBUG is enabled:
* Add an empty ao2 object to struct ast_module.
* Allocate ao2 object when the module is loaded.
* Perform an ao2_ref in each place where mod->usecount is manipulated.
* ao2_cleanup on module unload.
ASTERISK-24479 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4141/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Consider a scenario where Alice and Bob have an established dialog with each
other external to Asterisk. Bob decides to perform an attended transfer of
Alice to Asterisk. In this case, Alice will send an INVITE with Replaces
to Asterisk, where the Replaces specifies Bob's dialog with Asterisk. In this
particular scenario, Asterisk will complete the transfer, but - since Bob's
channel has had Alice masqueraded into it and is now a Zombie - a BYE
request will not be sent.
This patch fixes that issue by adding a new flag to chan_sip that tracks
whether or not we have an INVITE with Replaces. If we do, the flag is used
on the sip_pvt to ensure that a BYE request is sent, even if the channel has
been masqueraded away.
Review: https://reviewboard.asterisk.org/r/4362/
ASTERISK-22436 #close
Reported by: Eelco Brolman
Tested by: Jeremiah Gowdy, Kristian Høgh
patches:
asterisk-11-hangup-replaced-3.diff uploaded by Jeremiah Gowdy (License 6358)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies the ast_odbc_find_table function such that it only performs
a lookup of the requested table if the table is not already known. Prior to
this patch, a queries would be executed against the database even if the table
was already known and cached.
Review: https://reviewboard.asterisk.org/r/4405/
ASTERISK-24742 #close
Reported by: ibercom
patches:
patch.diff uploaded by ibercom (License 6599)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
scenarios when we are required to use SIPS URIs in Contact
headers. Asterisk's non-compliance with this could actually
cause calls to get dropped when communicating with clients
that are strict about checking the Contact header.
Both of the SIP stacks in Asterisk suffered from this issue.
This changeset corrects the behavior in chan_sip.
ASTERISK-24646 #close
Reported by Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/4346
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent security fix for OpenSSL broke DTLS negotiation for many
applications. This was caused by read ahead not being enabled when it
should be. While a commit has gone into OpenSSL to force read ahead
on for DTLS it may take some time for a release to be made and the
change to be present in distributions (if at all). As enabling read
ahead is a simple one line change this commit does that and fixes
the issue.
ASTERISK-24711 #close
Reported by: Jared Biel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request injection
can be performed given properly-crafted URLs.
Since Asterisk makes use of libcURL, and it is possible that users of Asterisk may
get cURL URLs from user input or remote sources, we have made a patch to Asterisk
to prevent such HTTP injection attacks from originating from Asterisk.
ASTERISK-24676 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/4364
AST-2015-002
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While running through some scenarios using chan_sip and tcp a problem would
occur that resulted in a flood of bad file descriptor messages on the cli:
tcptls.c:712 ast_tcptls_server_root: Accept failed: Bad file descriptor
The message is received because the underlying socket has been closed, so is
valid. This is probably happening because unloading of chan_sip is not atomic.
That however is outside the scope of this patch. This patch simply stops the
logging of multiple occurrences of that message.
ASTERISK-24728 #close
Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4380/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When refreshing (with a small expiration) a registration that was sent to
chan_sip the nonce would be considered stale and reject the registration.
What was happening was that the initial registration's "dialog" still existed
in the dialogs container and upon refresh the dialog match algorithm would
choose that as the "dialog" instead of the newly created one. This occurred
because the algorithm did not check to see if the from tag matched if
authentication info was available after the 401. So, it ended up assuming
the original "dialog" was a match and stopped the search. The old "dialog"
of course had an old nonce, thus the stale nonce message.
This fix attempts to leave the original functionality alone except in the case
of a REGISTER. If a REGISTER is received if searches for an existing "dialog"
matching only on the callid. If the expires value is low enough it will reuse
dialog that is there, otherwise it will create a new one.
ASTERISK-24715 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4367/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Starting and stopping conference recording more than once causes the
recording channels to be leaked. For v13 the channels also show up in the
CLI "core show channels" output.
* Reworked and simplified the recording channel code to use
ast_bridge_impart() instead of managing the recording thread in the
ConfBridge code. The recording channel's ref handling easily falls into
place and other off nominal code paths get handled better as a result.
ASTERISK-24719 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4368/
Review: https://reviewboard.asterisk.org/r/4369/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
All the other configuration options are case insensitive, so this one
should be too.
ASTERISK-24355 #close
Reported by: HZMI8gkCvPpom0tM
patches:
ast.patch uploaded by HZMI8gkCvPpom0tM (License 6658)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The MixMonitor m() option allows a recording to be pushed to a specific
voicemail mailbox. If the message is delivered to the mailbox's INBOX, however,
no MWI notification is currently raised.
This patch corrects the issue by properly calling notify_new_state from the
msg_create_from_file function. This will cause MWI to be triggered if the
message was placed in the mailbox's INBOX.
ASTERISK-24709 #close
Reported by: Gareth Palmer
patches:
app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On Debian based systems, the install_prereq tool uses a search command on
Debian that results in selecting both 64-bit and 32-bit packages. Besides the
waste of disk space, this can actually cause aptitude use 100% of memory on a
VM with 1GB of RAM as it tried to work out all of the 32-bit package
dependencies.
This patch filters out the 32-bit packages on a 64-bit machine, and leaves
32-bit machines alone.
ASTERISK-24048 #close
Reported by: Ben Klang
Tested by: Ben Klang, Matt Jordan
patches:
install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using ODBC or IMAP storage, temporary files created on the file system
must be disposed of using the DISPOSE macro. The DELETE macro will map to a
deletion function for the backend storage, but does not clean up any local
files created as a result of the operation.
When using voicemail with the operator and review options enabled, pressing
0 to enter the menu, followed by 1 to save the message, followed by any
other DTMF press to delete the message, will result in the temporary file
lingering on the file system.
This patch properly calls DISPOSE after the DELETE. This causes the local
file to be disposed of.
ASTERISK-24288 #close
Reported by: LEI FU
patches:
voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The mkpkgconfig script incorrectly concatenates Cflags options together. As an
example, the following:
Cflags: -I/usr/include/libxml2 -g3
Is instead generated as:
Cflags: -I/usr/include/libxml2-g3
This patch corrects the generation of Cflags in mkpkgconfig such that the
Cflags options are output correctly.
Review: https://reviewboard.asterisk.org/r/3707/
ASTERISK-23991 #close
Reported by: Diederik de Groot
patches:
fix_mkpkgconfig.diff uploaded by Diederik de Groot (License 6600)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
v11: If a channel redirect to a macro exten of a macro that is active
happens, the redirect location doesn't get executed. Instead the original
macro location is restored and gets reexecuted.
v13: An additional effect happens if a parked call times out to an
extension in the macro that parked the call then the macro is reexecuted
instead of the expected park return location.
* Made not restore the macro calling location on an
AST_SOFTHANGUP_ASYNCGOTO.
* Increased the locked channel range when setting up the macro execution
environment to cover things that should be done while the channel is
locked.
* Removed unnecessary NULL tests before calling ast_free() in
_macro_exec().
ASTERISK-23850 #close
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/4292/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The security event log uses a dynamic log level (SECURITY) that is registered
with the Asterisk logging core. Unfortunately, the syslog would ignore log
statements that had a dynamic log level associated with them. Because the
syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic
log entries sent to the syslog as logs with a level of NOTICE.
ASTERISK-20744 #close
Reported by: Michael Keuter
Tested by: Michael L. Young, Jacek Konieczny
patches:
asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the channel datastore associated with the usage of CURLOPT on a specific
channel is freed, the underlying structure holding the list of options is not
disposed of. This patch properly frees the structure in the datastore .destroy
callback.
ASTERISK-24672 #close
Reported by: Kristian Hogh
patches:
func_curl-memory-leak.diff uploaded by Kristian Hogh (License 6639)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change makes the T.38 negotiation timeout configurable via
't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously
hard coded to be 5000 milliseconds.
This change also handles T.38 switch failures by aborting the fax since
in the case where this can happen, both sides have agreed to switch to
T.38 and Asterisk is unable to do so.
Review: https://reviewboard.asterisk.org/r/4320/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the queues.conf.sample file to explicitly state which channel queue
variables are propagated to.
ASTERISK-24267
Reported by: Mitch Claborn
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The QUEUESTART log entry has historically acted like a fully booted event
for the queue_log file. When the QUEUESTART entry was posted to the log
was broken by the change made by ASTERISK-15863.
* Made post the QUEUESTART queue_log entry when Asterisk fully boots.
This restores the intent of that log entry and happens after realtime has
had a chance to load.
AST-1444 #close
Reported by: Denis Martinez
Review: https://reviewboard.asterisk.org/r/4282/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Given the following scenario:
* Three SIP phones (A, B, C), all communicating via a proxy with Asterisk
* A call is established between A and B. B performs a SIP attended transfer of
A to C. B sets the call on hold (A is hearing MOH) and dials the extension of
C. While phone C is ringing, B transfers the call (that is, what we typically
call a 'blond transfer').
* When the transfer completes, A hears the ringing of phone C, while B is idle.
In the SIP messaging for the above scenario, a REFER request is sent to
transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an
UPDATE request to phone C to update party information. This update is sent
directly to phone C, not through the intervening proxy. This has the unfortunate
side effect of providing route information, which is then set on the sip_pvt
structure for C. If someone (e.g. B) is trying to get the call back (through a
directed pickup), Asterisk will send a CANCEL request to C. However, since we
have now updated the route set, the CANCEL request will be sent directly to C
and not through the proxy. The phone ignores this CANCEL according to RFC3261
(Section 9.1).
This patch updates reqprep such that the route is not updated if an UPDATE
request is being sent while the INVITE state is INV_PROCEEDING or
INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent
to the correct location.
Review: https://reviewboard.asterisk.org/r/4279
ASTERISK-24628 #close
Reported by: Karsten Wemheuer
patches:
issue.patch uploaded by Karsten Wemheuer (License 5930)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The named ACL code incorrectly destroyed the config options information if loading
of the configuration file failed at startup. This would result in reloading
also failing even if a valid configuration file was put in place.
ASTERISK-23733 #close
Reported by: Richard Kenner
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the configuration section scheme of chan_dahdi.conf is used (keyword
dahdichan instead of channel) all setvar= options are completely ignored.
No variable defined this way appears in the created DAHDI channels.
* Move the clearing of setvar values to after the deferred processing of
dahdichan.
AST-1378 #close
Reported by: Guenther Kelleter
Patch by: Guenther Kelleter
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
