The config framework is supposed to be able to load configs that come from
multiple config files. The principle example is chan_sip's sip.conf and
users.conf. Unfortunately, it only does this correctly on initial load.
This patch causes the module's config to be reloaded entirely if any of
the config files change.
(closes issue ASTERISK-22009)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2859/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pjsip's message technology was being registered as 'sip', which was causing it
to not load due it conflicting with chan_sip's registered 'sip' technology for
messaging. It now registers as 'pjsip'. However, due to this change the "to"
field for outgoing pjsip messages need to be prefixed with 'pjsip:' instead of
'sip:'. Incoming messages to res_pjsip_messaging will automatically have their
"to" fields altered in order to accommodate the change. Outgoing messages also
handle changing it back to 'sip' before being sent so the pjsip library will
properly handle it.
(closes issue ASTERISK-22445)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2833/
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The endpoint option does not apply to communication with external entities. Rather,
the option is applied to all communications with the endpoint. The external_media_address
transport configuration option may override the endpoint option if it turns out that
we are going to be communicating with an external entity.
Two things of note:
1) I have not updated the XML documentation. This is being taken care of by Rusty as part
of his work on issue ASTERISK-22405
2) This commit is likely to cause testsuite failures since there are tests that use the
external_media_address endpoint option, and they will need to be changed over. Well, I'm
planning to get that updated ASAP after this commit.
(closes issue ASTERISK-22528)
reported by Rusty Newton
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The remote console continued to have issues with its output. In this case CLI
command output would either not show up (if verbose level = 0) or would contain
verbose prefixes (if verbose level > 0) once log messages were sent to the
remote console. The fix now now adds verbose prefix data to all new lines
contained in a verbose log string.
(closes issue ASTERISK-22450)
Reported by: David Brillert
(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2825/
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Confbridge would not properly tear down an empty conference bridge when all
users were kicked via end_marked=yes and at least one user was also set to
wait_marked. This occurred because while end_marked users were being kicked
and at least one was also set to wait_marked then the leave wait_marked handler
would be called on that user, but there would be no waiting user (still
considered active). The waiting users would decrement and now be negative. The
conference would remain, but be put into an inactive state. The solution was
to move from the active list to the wait list, those users with wait_marked set
right before kicking. This allows both the active and wait users to decrement
correctly and the confbridge to tear down properly.
A crashed also occurred when trying to list the specific conference from the CLI.
This happened because the conference specified was invalid. Since the
conference properly tears down now there is no way to reference it thus
alleviating the crash as well.
(closes issue ASTERISK-21859)
Reported by: Chris Gentle
Review: https://reviewboard.asterisk.org/r/2848/
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You cannot use a function pointer variable with an external function from
another dynamically loaded module because data variables are always
resolved even with RTLD_LAZY.
* Added wrapper functions for ast_ari_validate_int() and
ast_ari_validate_string() to use instead for the function pointer
variable.
(closes issue ASTERISK-22457)
Reported by: David M. Lee
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Fixes regression introduced by -r374096.
* Made res_speech.export.in export ast_* symbols instead of specific
functions.
* Made app_speech_utils.c declare that it is dependent upon res_speech.
(issue ASTERISK-17136)
Reported by: Richard Kenner
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When a new IAX2 client registers, the astdb database is updated with the
value of minregexpire defined in iax.conf instead of using the expiry time
that is provided by the client. The provided expiry time of the client is
updated after inserting the astdb entry. As a consequence, restarting or
reloading asterisk creates clients whose registration may expire before
they reregister. The clients are therefore unavailable after minregexpire
seconds until they reregister.
* Move updating of the expiry time to before inserting into the astdb.
(closes issue ASTERISK-22504)
Reported by: Stefan Wachtler
Patches:
chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler
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Some channels exist merely as an implementation detail in Asterisk, such as
ConfBridge's announcer/recorder channels. These channels should never be
exposed to the outside world, or to interfaces that report on Asterisk. We
already filter out such channels in snapshot processing; however, we failed to
filter out bridge related messages that involved these channels.
This patch filters out bridge related messages that are for such channels. This
prevents a spurious WARNING message from being displayed when those channels
move in and out of bridges.
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The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
(closes issue ASTERISK-22072)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2845/
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With this change, if no realm is specified in an outbound auth
section, then we will simply match the realm that was present
in the 401/407 challenge.
(closes issue ASTERISK-22471)
Reported by George Joseph
(closes issue ASTERISK-22386)
Reported by Rusty Newton
Patches:
outbound_auth_realm_v4.patch uploaded by George Joseph (License #6322)
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This patch uses PJSIP's pj_log_set_log_func() to forward PJSIP's log
messages to Asterisk's logger. This is done in a new module:
res_pjsip_log_forwarder.so.
This patch sets defaultenabled on the existing res_pjsip_logger.so to
no, since logging every SIP packet seems a bit odd to do by default, and
is (hopefully) less necessary with regular PJSIP logging.
It also removes res_rtp_asterisk's disabling of PJSIP logging.
(closes issue ASTERISK-22360)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2830/
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When I moved the ARI WebSocket from /ws to /ari/events, I added code to
allow a WebSocket to connect without specifying the subprotocol if
there's only one subprotocol handler registered for the WebSocket.
Naively, I coded it to always respond with the subprotocol in use.
Unfortunately, according to RFC 6455, if the server's response includes
a subprotocol header field that "indicates the use of a subprotocol that
was not present in the client's handshake [...], the client MUST _Fail
the WebSocket Connection_.", emphasis theirs.
This patch correctly omits the Sec-WebSocket-Protocol if one is not
specified by the client.
(closes issue ASTERISK-22441)
Review: https://reviewboard.asterisk.org/r/2828/
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* One bug fix. Made the synopsis for "type" to accurate.
* changing the usage of "IP-domains" to "IP addresses"
* clarifying the usage for the options, by adding a relevant description for
each
* modified other areas of the XML help for clarity, such as the module
description and a few synopsis changes here and there. See the patch.
(issue ASTERISK-22458)
(closes issue ASTERISK-22458)
Reported By: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2823/
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The masquerade super test is failing on v12 with high fence violations and
crashing. The fence violations are showing that party id allocated memory
strings are somehow getting corrupted in the
bridge_reconfigured_connected_line_update() function. The invalid string
values happen to be the freed memory fill pattern.
After much puzzling, I deduced that the
bridge_reconfigured_connected_line_update() is copying a string out of the
source channel's caller party id struct just as another thread is updating
it with a new value. The copying thread is using the old string pointer
being freed by the updating thread. A search of the code found the
unreal_colp_redirect_indicate() routine updating the caller party id's
without holding the channel lock.
A latent bug in v1.8 and v11 hatched in v12 because of the bridging and
connected line changes. :)
(issue ASTERISK-22221)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2839/
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Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:
1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.
Also added an SDP when an update is sent out.
(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
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Richard Mudgett
Kinsey Moore
Jonathan Rose
Joshua Colp
Kevin Harwell
Matthew Jordan
Michael L. Young
Mark Michelson
David M. Lee
Rusty Newton
Russell Bryant