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load"
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header"
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stdout"
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When transfering to a URI without an extension, ensure that the
s extension of the dialplan is entered
ASTERISK-26869 #close
Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525
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A caller can leave the Queue() application after being bridged with a
member in a few ways:
* Caller or member hangup
* Caller is transferred somewhere else (blind or atx)
* Caller is externally redirected elsewhere
The first 2 scenarios are currently handled by subscribing to stasis
messages, but the 3rd is not explicitly covered. If a caller is
redirected away from the Queue() application, the member who was last
bridged with that caller will remain in an "In use" state until the
caller hangs up.
This patch adds handling of the caller leaving the queue via
redirection. We monitor the caller-member bridge, and if the caller is
the one that leaves, we treat it the same as we would a caller hangup.
ASTERISK-26400 #close
Reported by: Etienne Lessard
Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334
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This change ensures that if no header_match option is set on an
identify an error message is not output stating the option is set
to an invalid value.
ASTERISK-26863
Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a
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This patch adds a new features to the endpoint identifier module,
'match_header'. When set, inbound requests are matched by a provided SIP
header: value pair. This option works in conjunction with the existing
'match' configuration option, such that if any 'match*' attribute
matches an inbound request, the request is associated with the specified
endpoint.
Since this module now identifies by more than just IP address,
appropriate renaming of the module and/or variables can be done in a
non-release branch.
ASTERISK-26863 #close
Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453
(cherry picked from commit 30f52d79d7fc9ab0b628bef2b61ea515413795a2)
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nonexistent item"
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Forgetting to indicate an exten is a pattern can cause a crash if the
"pattern" has a character set range. e.g., "9999[3-5]" The crash is due
to a buffer overwrite because the '-' exten eye-candy wasn't removed as
expected and overran the allocated space.
The buffer overwrite is fixed two ways in this patch.
1) Fix ext_strncpy() to distinguish between pattern and non-pattern
extens. Now '-' characters are removed when they are eye-candy and not
when they are part of a pattern character set. Since the function is
private to pbx.c, the return value now returns the number of bytes written
to the destination buffer instead of the strlen() of the final buffer so
the callers that care don't need to add one.
2) Fix callers to ext_strncpy() to supply the correct available buffer
size of the destination buffer.
ASTERISK-26668
Change-Id: I555d97411140e47e0522684062d174fbe32aa84a
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When using a non-dynamic peer address, build_peer() invalidates the
peer address structure by setting the address family to unspecified.
However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup()
will not amend the peer address if the cache is still valid, resulting
in peer connectivity failures.
To fix this, we call ast_dnsmgr_refresh() instead.
ASTERISK-26865
Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082
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In some scenarios, such as when there may not be a terminal (such as
inside a Docker container), curl will apparently direct the progress bar
to stdout. This can cause extra data to be appended to a file curl'd
down to stdout, resulting in md5 verification failures.
This patch removes the progress bar, and tells curl to download the file
silently.
ASTERISK-26872 #close
Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c
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This patch updates the documenation in hep.conf.sample to better specify
how the various HEP modules interact.
ASTERISK-26717 #close
Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124
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During module loading of func_devstate, Asterisk emits the current
device state of all Custom device states currently stored in the AstDB.
This was erroneously including a new line character ('\n') to the end of
the device state, causing two new lines to be emitted in
DeviceStateChange AMI events.
Note that this only happened for those device state changes that
occurred during startup. Regular device state changes for Custom device
states are handled elsewhere, and did not have the newline.
ASTERISK-26643 #close
Reported by: Roman Bedros
Tested by: Matt Jordan
patches:
ami_devstate.diff uploaded by Roman Bedros (License 6842)
Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93
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This patch demotes the ERROR message that is displayed when a
nonexistent item is removed from the Stasis cache. The genesis of this
demotion is due to chan_sip's realtime peers and their interaction with
Asterisk's core ast_endpoint code, but ostensibly it could happen from
other channel drivers as well.
Since Mark Michelson already did an excellent job of explaining on this
issue, it is quoted here for posterity:
"Internally, when a realtime peer is retrieved, Asterisk creates an
ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
destroyed as well. Part of the destruction of the ast_endpoint involves
clearing the Stasis cache of all information about that endpoint. The
problem here is that the act of creating the ast_endpoint is not enough
to actually put any information in the Stasis cache. Instead, something
has to happen, such as a state change, in order for the Stasis cache to
have any information about that endpoint. When a device registers,
chan_sip creates an ast_endpoint structure, processes the REGISTER, and
then destroys the ast_endpoint. When the ast_endpoint is destroyed,
there is nothing to destroy in the Stasis cache, so an error message is
emitted. When you use rtcachefriends, ast_endpoint structures persist
for the lifetime of the module and so you do not see this error
message."
ASTERISK-25237 #close
Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70
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Tabs > spaces. Always.
Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1
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When querying for PJSIP specific information using the dialplan
function CHANNEL() it is possible that the underlying session
will no longer have a channel associated with it. This is
most likely to occur when the RTCP HEP module attempts to get
the channel name. If this happens then a crash will occur.
This change just adds a check that the channel exists on the
session before querying it.
ASTERISK-26857
Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01
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Bundled pjproject should now only rebuild if one of the menuselect
"Compiler Flags" options changes.
Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43
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* cli_commands.c Fixed CLI output
ASTERISK-26822 #close
Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01
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* res_musiconhold.c: Ensure the general section is not treated as
a moh class.
ASTERISK-26353 #close
Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d
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Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def
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Set a variable on the channel that indicates which attempt number we
are currently performing to allow for attempt-specific behavior.
ASTERISK-26568 #close
Reported by: Roman Shubovich
Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89
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This change adds a PJSIP patch (which has been contributed upstream)
to allow the registration of IPv6 transport types.
Using this the res_pjsip_transport_websocket module now registers
an IPv6 Websocket transport and uses it for the corresponding
traffic.
ASTERISK-26685
Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647
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* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.
ASTERISK-24562 #close
Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe
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When doing some WebRTC testing, I found that the websocket would
disconnect whenever I attempted to place a call into Asterisk. After
looking into it, I pinpointed the problem to be due to the iostreams
change being merged in.
Under certain circumstances, a call to ast_iostream_read() can return a
negative value. However, in this circumstance, the websocket code was
treating this negative return as if it were a partial read from the
websocket. The expected length would get adjusted by this negative
value, resulting in the expected length being too large.
This patch simply adds an if check to be sure that we are only updating
the expected length of a read when the return from a read is positive.
ASTERISK-26842 #close
Reported by Mark Michelson
Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab
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When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.
ASTERISK-26841
Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099
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This change adds a few things to facilitate stream topology changing:
1. Control frame types have been added for use by the channel driver
to notify the application that the channel wants to change the stream
topology or that a stream topology change has been accepted. They are
also used by the indicate interface to the channel that the application
uses to indicate it wants to do the same.
2. Legacy behavior has been adopted in ast_read() such that if a
channel requests a stream topology change it is denied automatically
and the current stream topology is preserved if the application is
not capable of handling streams.
Tests have also been written which confirm the multistream and
non-multistream behavior.
ASTERISK-26839
Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9
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* say.c Changed 'digits/and' to 'vm-and' for en_GB
ASTERISK-26598 #close
Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe
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Per the linked issue, we aren't checking the buffer filled by fgets()
to determine if it contains a newline, so we will fail to correctly
parse the trailing portion of a long line.
This patch increases the buffer size from 256 to 1024, and skips any
line that exceeds that length, logging a warning in the process.
ASTERISK-17067 #close
Reported by: Dave Olszewski
Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0
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* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension. Ran into this when
developing a testsuite test. The AMI event ExtensionStatus came out with
the hint header value containing garbage. The AMI event PresenceStatus
also had the same issue.
* manager.c:action_extensionstate() no need to completely initialize the
hint[]. Only initialize the first element.
* pbx.c:ast_add_hint() Remove unnecessary assignment.
* chan_sip.c: Eliminate an unneeded hint[] local variable. We only care
about the return value of ast_get_hint() there.
Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b
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According to the RFC[1] WSS should only be used in the Via header
for secure Websockets.
* Use WSS in Via for secure transport.
* Only register one transport with the WS name because it would be
ambiguous. Outgoing requests may try to find the transport by name and
pjproject only finds the first one registered. This may mess up unsecure
websockets but the impact should be minimal. Firefox and Chrome do not
support anything other than secure websockets anymore.
* Added and updated some debug messages concerning websockets.
* security_events.c: Relax case restriction when determining security
transport type.
* The res_pjsip_nat module has been updated to not touch the transport
on Websocket originating messages.
[1] https://tools.ietf.org/html/rfc7118
ASTERISK-26796 #close
Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12
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