When two users entered a new conference simultaneously, one of the callers
hears MOH. This happened if two unmarked users entered simultaneously and
also if a waitmarked and a marked user entered simultaneously.
* Created a confbridge internal MOH API to eliminate the inlined MOH
handling code. Note that the conference mixing bridge needs to be locked
when actually starting/stopping MOH because there is a small window
between the conference join unsuspend MOH and actually joining the mixing
bridge.
* Created the concept of suspended MOH so it can be interrupted while
conference join announcements to the user and DTMF features can operate.
* Suspend any MOH until the user is about to actually join the mixing
bridge of the conference. This way any pre-join file playback does not
need to worry about MOH.
* Made post-join actions only play deferred entry announcement files.
Changing the user/conference state during that time is not protected or
controlled by the state machine.
(closes issue ASTERISK-20606)
Reported by: Eugenia Belova
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2232/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Review the syntax of the 'skinny debug' command to show more than
just 'show' for options to 'skinny debug' command.
(closes issue ASTERISK-20789)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-debug.diff uploaded by snuffy (license 5024)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk now includes Min-SE in outbound INVITEs when the value is not
90 (the default) and session timers are not disabled. This has the
effect of Asterisk following RFC4028 more closely with regard to 422
responses and preventing situations in which Asterisk would be forced
to temporarily accept a call to tear it down based on a Session-Expires
below the locally configured Min-SE.
(issue SWP-5051)
Review: https://reviewboard.asterisk.org/r/2222/
Reported-by: Kinsey Moore
Patch-by: Kinsey Moore
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The issue comes from the fact that transfers may perform
a redirecting update on a channel. The issue is that lock
inversion between the channel and its tech_pvt occurs since
the channel lock is released during the transfer process.
The fix is to move when the redirecting update occurs to a
place where neither the tech_pvt or the channel is locked so
that the two can be locked in the proper order.
(closes issue ASTERISK-20708)
reported by Mark Michelson
patches:
ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049)
Tested by:
Tim Ringenbach at Asteria Solutions Group
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent memory leak fix in main/cli.c causes an ast_cli_entry's command
field to be freed and NULLed if ast_cli_register() fails. res_clialiases
was ignoring the return value of ast_cli_register() and was then passing
the NULL command off to a a hash function. This resulted in a crash.
The fix is not to ignore the erroneous return value. If ast_cli_register()
fails, then we do not continue trying to process the current alias.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made ast_unregister_indication_country() unlink the found tone zone
before selecting a new default_tone_zone to make it impossible to select
the tone zone being unregistered again.
* Ringcadence is no longer parsed twice in store_config_tone_zone().
* Cleanup CLI commands and destroy default_tone_zone on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
Modified
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using res_fax_digium, the T.38 CED tone was not being provided
properly which would cause some incoming faxes to fail. This was not an
issue with res_fax_spandsp since it does not strictly honor the
send_ced flag and sends the CED tone whenever receiving a T.38 fax.
(closes issue FAX-343)
Reported-by: Benjamin Tietz
Patch-by: Kinsey Moore
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In r306010 "Asterisk media architecture conversion - no more format
bitfields", the logic for incrementing encoders and decoders when
opening transcoder channels was changed without making the corresponding
change when decrementing encoder / decoder channels. The result being
that when a channel was destroyed, codec_dahdi couldn't properly tell if
it was an encoder or decoder, and the default case is to assume it was a
decoder.
This could result in negative numbers for decoders in use like in:
VOIP6*CLI> transcoder show
2/-2 encoders/decoders of 92 channels are in use.
(closes issue ASTERISK-19921)
Patch-by: Shaun Ruffell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made destroy_conference_bridge() destroy a missed ast_mutex_t and ast_cond_t.
* Made join_conference_bridge() init the ast_mutex_t's and ast_cond_t so
destroy_conference_bridge() can destroy them unconditionally.
* Made join_conference_bridge() abort if the new conference could not be
added to the conferences container.
* Made leave_conference() discard any post-join actions if
join_conference_bridge() had to abort early.
* Made the join_conference_bridge() diagnostic messages better describe
what happened.
* Renamed leave_conference_bridge() to leave_conference() and made it only
take a conference user pointer. The conference pointer was redundant.
* Made conf_bridge_profile_copy() use struct copy instead of memcpy().
* No need to lock the conference in start_conf_record_thread() since all
of the callers already have it locked.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the CLI command 'manager show event' was run incorrectly and its usage
instructions returned, a reference to the event container was leaked. This
would prevent the container from being reclaimed when Asterisk exits. We now
properly decrement the count on the ao2 object using the nifty RAII_VAR macro.
Thanks to Russell for helping me stumble on this, and Terry for writing that
ridiculously helpful macro.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During the TLS re-work in chan_sip some TLS specific code was moved
into a separate function. This function operates on a copy of the
incoming SIP request. This copy was never deinitialized causing a
memory leak for each request processed.
This function is now given a SIP request structure which it can use
to copy the incoming request into. This reduces the amount of memory
allocations done since the internal allocated components are reused
between packets and also ensures the SIP request structure is
deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763)
Reported by: deti
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made func_confbridge_helper() allow an empty value when setting options.
You previously could not Set(CONFBRIDGE(user,pin)=) and clear the
configured pin from the dialplan.
* Made func_confbridge_helper() handle its datastore better if multiple
threads attempt to set the first CONFBRIDGE option value on the channel.
* Made the func_confbridge_helper() only output one diagnostic message
concerning the option.
* Made the bridge video_mode able to repeatedly change in the config file
and CONFBRIDGE dialplan function. The video_mode option values are an
enum and not independent of each other.
* Made handle_cli_confbridge_show_bridge_profile() better handle the
video_mode option.
* Simplified datastore handling code in conf_find_user_profile() and
conf_find_bridge_profile().
(closes issue ASTERISK-20655)
Reported by: Birger "WIMPy" Harzenetter
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When setting up an RTP instance the RTCP portion of the instance
keeps a reference to the instance itself. In order to release this
reference and stop RTCP the stop API call must be called before
destroying the instance.
(closes issue ASTERISK-20751)
Reported by: joshoa
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Based on feedback from numerous individuals this patch tweaks incoming calls
to first look for an extension with the name of the endpoint. If no such extension
exists the call will silently fall back to the "s" extension as it previously
did.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The principal behind this patch is simple. During a transfer,
we manipulate channels that are owned by a separate thread than
the one we currently are running in, so it makes sense that we
need to grab a reference to the channels so that they cannot
disappear out from under us.
In the wild, crashes were sometimes seen when the transferring
party would hang up the call before the transfer target answered
the call. The most common place to see the crash occur was when
attempting to send a connected line update to the transferer
channel.
(closes issue ASTERISK-20226)
Reported by Jared Smith
Patches:
ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
Tested by: Jared Smith
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For 1.8, 10, 11 and trunk we are are improving the code readability.
For 11 and trunk, auto nat detection was added. The natdetected flag was being
set to 1 when the host address in the VIA header did not specifiy a port. This
patch fixes this by setting the port on the temporary sock address used to
SIP_STANDARD_PORT in order for the sock address comparison to work properly.
(closes issue ASTERISK-20724)
Reported by: Michael L. Young
Patches:
asterisk-20724-set-port-v2.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2206/
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Websocket by default doesn't return an ast_str for the payload received. When
converting it to an ast_str on chan_sip the last character was being omitted,
because ast_str functions expects that the given length includes the trailing
0x00. payload_len only has the actual string length without counting the
trailing zero.
For most cases this passed unnoticed as most of SIP messages ends with \r\n.
(closes issue ASTERISK-20745)
Reported by: Iñaki Baz Castillo
Review: https://reviewboard.asterisk.org/r/2219/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
