Age | Commit message (Collapse) | Author |
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The context/extension in a CDR is generally considered the destination of a
call. When looking at a 2-party call CDR, users will typically be presented
with the following:
context exten channel dest_channel app data
default 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20
However, if the Dial actually takes place in a Macro, the current behaviour
in 12 will result in the following CDR:
context exten channel dest_channel app data
macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20
The same is true of a GoSub:
context exten channel dest_channel app data
subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20
This generally makes the context/exten fields less than useful.
It isn't hard to preserve these values in the CDR state machine; however, we
need to have something that informs us when a channel is executing a
subroutine. Prior to this patch, there isn't anything that does this.
This patch solves this problem by adding a new channel flag,
AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a
Macro or a GoSub. The CDR engine looks for this value when updating a Party A
snapshot; if the flag is present, we don't override the context/exten on the
main CDR object. In a funny quirk, executing a hangup handler must *not* abide
by this logic, as the endbeforehexten logic assumes that the user wants to see
data that occurs in hangup logic, which includes those subroutines. Since
those execute outside of a typical Dial operation (and will typically have
their own dedicated CDR anyway), this is unlikely to cause any heartburn.
Review: https://reviewboard.asterisk.org/r/3962/
ASTERISK-24254 #close
Reported by: tm1000, Tony Lewis
Tested by: Tony Lewis
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This patch fixes an issue where CDRs would get stuck generating an infinite
number of CDRs, eventually crashing Asterisk (and consuming a lot of memory
along the way).
When a channel enters into a multi-party bridge, the CDR engine creates
mappings of each participant to each other participant, picking the 'A' party
as it goes. So, if we have four channels in a multi-party bridge (Alice, Bob,
Charlie, Denise), we would have something like:
Alice => Bob
Alice => Charlie
Alice => Denise
Bob => Charlie
Bob => Denise
Charlie => Denise
This works fine when participants enter the bridge a single time.
When a participant leaves a bridge, the CDRs for that channel are transitioned
to a finalized state.
The bug occurs if Bob rejoins. When the CDR engine creates mappings between the
channels, it walks through all the participants currently in the bridge, and
realizes that no one in the bridge can create a CDR with the channel (Bob).
As such it creates a new CDR for the candidate and appends it to that
candidate's chain. Unfortunately, on this particular code path, it doesn't
stop traversing the candidate's chain. Since we just added ourselves to the
chain, this causes the loop to keep going, constantly adding new CDRs.
This patch makes it so the engine bails when it creates a CDR match in this
case.
Review: https://reviewboard.asterisk.org/r/3964/
ASTERISK-24241 #close
Reported by: Deepak Singh Rawat
Tested by: Deepak Singh Rawat
ASTERISK-24208
Reported by: Frankie Chin
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* The CHANNEL() audionativeformat, videonativeformat, audioreadformat, and
audiowriteformat now need locking since the media format rework when
accessing the channel's format pointers.
* Increased the buffer size for CHANNEL() audionativeformat and
videonativeformat output strings since the allow=all can be a lengthy
list.
* Tweaked the CHANNEL() XML documentation for secure_bridge_signaling,
secure_bridge_media, and state.
* Ensured the output buffer is initialized for secure_bridge_signaling and
secure_bridge_media.
* Made use the locked_copy_string() macro instead of inlining it for trace
and checkhangup.
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Adds an option to the dial API that marks an outgoing dial as replacing the dialing channel for the purpose of propagating accountcode. When it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved channels with ast_channel_req_accountcodes.
Review: https://reviewboard.asterisk.org/r/3968/
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NULL call IDs were meant to appear as '(none)' but instead were showing
the contents of an uninitialized character buffer.
ASTERISK-24223
Review: https://reviewboard.asterisk.org/r/3979/
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* In ast_state_chan2dev() use ARRAY_LEN() instead of a sentinel value in
chan2dev[].
* Fix some comments in chan_iax2.c.
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This corrects a situation where menuselect can incorrectly enable a
module by default that has defaultenabled set to "no" and has
failed/non-selected dependencies. The bug is due to an inverted test
when checking for whether the given module should be set to enabled by
default on load.
Review: https://reviewboard.asterisk.org/r/3975/
Reported by: John Bigelow
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Review: https://reviewboard.asterisk.org/r/3969/
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The code for changing the Contact header wrongly assumed that the Contact
would always contain a URI. This is incorrect.
ASTERISK-24271
Reported by: Dafi Ni
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media has been negotiated.
Testsuite tests will occasionally fail because on reception of a 200 OK SIP response,
an AST_CONTROL_ANSWER frame is queued prior to when media has finished being
negotiated. This is because session supplements are called into before PJSIP's
inv_session code has told us that media has been updated. Sometimes the queued answer
frame is handled by the PBX thread before the ensuing media negotiations occur, causing
a test failure.
As it turns out, there is another place that session supplements could be called into, which is
after media has finished getting negotiated. What this commit introduces is a means for session
supplements to indicate when they wish to be called into when handling an incoming SIP response.
By default, all session supplements will be run at the same point that they were prior to this
commit. However, session supplements may indicate that they wish to be handled earlier than
normal on redirects, or they may indicate they wish to be handled after media has been negotiated.
In this changeset, two session supplements have been updated to indicate a preference for when
they should be run: res_pjsip_diversion executes before handling redirection in order to get
information from the Diversion header, and chan_pjsip now handles responses to INVITEs after
media negotiation to fix the race condition mentioned previously.
ASTERISK-24212 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/3930
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Internal channels don't have CDRs. Querying the CDR engine for their variables
will make it cranky.
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When ARI manipulates a bridge, it generally doesn't care what the mixing
technology is. Operations on a bridge initiated through ARI should perform
their action in generally the same way, regardless of the bridge's mixing
technology. While the mixing technology may determine how media flows to
channels, the actual operations on a bridge themselves should be the same.
Currently, this isn't the case with holding bridges. When a channel joins
without a role, MoH is started on that channel automatically. Subsequent bridge
operations that would stop MoH would fail (as there is no Announcer channel
playing MoH to the bridge). Starting MoH on the bridge will also create two
MoH streams: one from the MoH being played on the participant channel, and one
from the announcer channel. From the perspective of ARI users, this is
counter-intuitive - I would not expect MoH to be started for me. The mixing
technology determines how media is shared between participants, not the
application experience.
This patch does the following:
* The Stasis bridge class now inspects channels as they are going into a
bridge. If the bridge has a holding capability, and the channel has no
roles, we give it a participant role and mark the default behaviour to have
no entertainment. This allows addChannel operations to continue to set a
participant role with an entertainment option if it felt like it (or could
do it).
* The music on hold channel is now Stasis approved (tm)
Review: https://reviewboard.asterisk.org/r/3929/
ASTERISK-24264 #close
Reported by: Samuel Galarneau
Tested by: Samuel Galarneau
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The ConfbridgeList event doesn't include how long the user has been a
member of the conference. This patch adds Duration (seconds) which
is based on user->chan->answertime.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3955/
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A WaitEvent issued via an http session isn't respecting eventfilters defined
for the user. I just added a match_filter to the predicate that controls
astman_append.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3958/
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This patch adds a manpage for the smsq utility. Note that this is one of
the patches the Debian distro applies for the Asterisk project, as per
ASTERISK-24191.
Review: https://reviewboard.asterisk.org/r/3895/
ASTERISK-24171 #close
Reported by: Jeremy Laine
patches:
smsq.8 uploaded by Jeremy Laine (License 6561)
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This patch adds a manpage for the aelparse utility. Note that this is one of
the patches the Debian distro applies for the Asterisk project, as per
ASTERISK-24191.
Review: https://reviewboard.asterisk.org/r/3896/
ASTERISK-24171 #close
Reported by: Jeremy Laine
patches:
aelparse.8 uploaded by Jeremy Laine (License 6561)
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message was being triggered on configuration reload.
This patch changes that case to just return instead.
Review: https://reviewboard.asterisk.org/r/3953/
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The UniMRCP project distributes Asterisk modules that integrate Asterisk with
UniMRCP, and other Asterisk users use the UniMRCP library as well.
Unfortunately, the UniMRCP license is Apache 2.0, which per the Free Software
Foundation, is not a compatible license with the GPLv2.
"Please note that this license is not compatible with GPL version 2, because it
has some requirements that are not in that GPL version. These include certain
patent termination and indemnification provisions. The patent termination
provision is a good thing, which is why we recommend the Apache 2.0 license for
substantial programs over other lax permissive licenses."
On the other hand, UniMRCP is a great project and we'd like to let people use
it with Asterisk.
This patch updates the LICENSE text to allow users to link Asterisk with
UniMRCP and distribute the resulting binaries.
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The reporter on the issue found some issues when upgrading from version 10 to 11
on 55 hosts.
Two situations that can occur with dynamic registrations.
1. With dnsmgr disabled, if the host is not resolvable we are not trying to
resolve the host again when it is time to attempt to register again. This
results in never registering to the host.
2. With dnsmgr enabled, when the host is temporarily not resolvable the
address is set to 0.0.0.0:0 and then when the host is resolvable the port
is not being restored and stays set to 0.
This patch resolves these two issues by:
* Storing the hostname so that it can be used for resolving with DNS.
* Resolve the hostname on the next scheduled attempt to register.
* Storing the port used to reach the host so that when the hostname is
resolvable again, we can set the port again if the port is still unset after
looking up the host.
ASTERISK-23767 #close
Reported by: David Herselman
Tested by: David Herselman, Michael L. Young
Patches:
asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3856/
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Users now have the ability to bind the rtpengine instance to a specific IP
address. For example, you want chan_sip (call control) on eth0 but rtp (media)
on eth1.
ASTERISK-24280 #close
Reported by: Paul Belanger
Tested by: Paul Belanger
Review: https://reviewboard.asterisk.org/r/3952/
Patches:
rtpengine.diff uploaded by Paul Belanger
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A misunderstanding of how the scheduler worked caused further batched notifications
beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after
the batched notification is sent. This way, further notifications can be scheduled
when they arise.
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* Fix off nominal ref leak in find_or_create_contact_status().
* Add missing NULL check of status in update_contact_status() and
init_start_time().
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Currently there's no way to tell if a user is an admin or not when receiving
the join, leave, mute, unmute and talking events. This patch adds that
capability.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3950/
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This allows the callerid parsing function to handle malformed input
strings and strings containing escaped and unescaped double quotes.
This also adds a unittest to cover many of the cases where the parsing
algorithm previously failed.
Review: https://reviewboard.asterisk.org/r/3923/
Review: https://reviewboard.asterisk.org/r/3933/
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Kick, mute and unmute were a little inconsistent in their handling of channel
targets. This patch cleans that up by insuring they all handle the 'all'
target consistently and adds the 'participants' target which acts on
non-admins. Documentation for kick was also cleaned up as it never
supported partial channel names.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3944/
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When scheduled tasks run, they are removed from the heap (or hashtab).
When a scheduled task is deleted, if the task can't be found in the
heap (or hashtab), an assertion is triggered. If DO_CRASH is enabled,
this assertion causes a crash.
The problem is, sometimes it just so happens that someone attempts
to delete a scheduled task at the time that it is running, leading
to a crash. This change corrects the issue by tracking which task
is currently running. If that task is attempted to be deleted,
then we mark the task, and then wait for the task to complete.
This way, we can be sure to coordinate task deletion and memory
freeing.
ASTERISK-24212
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/3927
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* Clear the channel music_state pointer before destroying the music_state
object for safety.
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Restore code removed by https://reviewboard.asterisk.org/r/3536/ that
introduced a regression that prevents MOH from restarting were it left off
the last time.
ASTERISK-24019 #close
Reported by: Jason Richards
Patches:
jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/3928/
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In order to alter the Contact header on in-dialog requests and responses the
Websocket module must be attached on outgoing INVITEs. The Contact header is
modified so that the PJSIP transport layer can find and use the existing
Websocket connection based on the source IP address, port, and transport.
ASTERISK-24143 #close
Reported by: Aleksei Kulakov
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The packet structure used to receive messages was using the transport
pool. This meant that for each parsing the pool would grow accordingly.
Since memory can not be reclaimed without resetting it this would
cause the memory pool to grow and grow.
This change uses a specific memory pool for the packet structure and
resets it to a fresh state after the message has been received and
handled.
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This change enforces the transport in the Contact header for Websocket clients.
Previously a client may provide a transport of 'ws' when it is actually using
a transport of 'wss'. This would cause outgoing calls to fail as the existing
connection could not be found.
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This code originally worked around an issue within res_rtp_asterisk itself.
The wrong socket was being used for the STUN check for RTCP, causing the
port to be the same as RTP. This was subsequently fixed and the RTCP port
provided for the ICE candidate is correct and does not need to be incremented.
ASTERISK-23997 #close
Reported by: Badalian Vyacheslav
Patches:
plus1.diff submitted by Badalian Vyacheslav (license 5249)
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We need to unlock the audiohook before trying to lock
the channel, since the correct locking order is channel
then audiohook.
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ASTERISK-24147 #close
Reported by: Edvin Vidmar
Review: https://reviewboard.asterisk.org/r/3908/
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Remove unneeded code that writes to the wrong file location in an obsolete
format.
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Using the hostname in the SDP origin line may not satisfy the requirement
of RFC 4566 that we use a FQDN or IP address. This change has us use the
same information from the SDP connection line if possible. If not possible,
we'll use the configured media address. And if that's not possible, we use
the result of a PJLIB call to get the IP address of ourself.
ASTERISK-23994 #close
Reported by Private Name
Review: https://reviewboard.asterisk.org/r/3925
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non-Stasis bridge.
Because of the departable state of channels that enter Stasis bridges, Stasis has to
take responsibility for directing the channel to its intended after-bridge destination
if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures
that when such a move occurs, when the channel leaves the bridging system, any after
bridge gotos are honored.
Review: https://reviewboard.asterisk.org/r/3920
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Due to a faulty function for debugging reference decrementing, it was possible
to reduce the refcount on the wrong object if two moh classes of the same name
were in the moh class container.
(closes issue ASTERISK-22252)
Reported by: Walter Doekes
Patches:
18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182)
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Prior to this change, the Remote-Party-ID header took the position of
"If caller name and number are not explicitly allowed, then they are private"
and P-Asserted-Identity took the position of
"Caller name and number are only private if marked explicitly so"
Now both mechanisms of conveying party identification use the former approach.
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If a user does not provide a port in the fromdomain setting, chan_sip will set
the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will
then get used unilaterally in certain places. This causes issues with TLS,
where the default port is expected to be 5061.
This patch modifies chan_sip such that fromdomainport is only used if it is
not the standard SIP port; otherwise, the port from the SIP pvt's recorded
self IP address is used.
Review: https://reviewboard.asterisk.org/r/3893/
ASTERISK-24178 #close
Reported by: Elazar Broad
patches:
fromdomainport_fix.diff uploaded by Elazar Broad (License 5835)
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When issuing a POST /channels/{channel_id}/play on a channel that is not
yet answered, ARI is supposed to:
* Queue up an AST_CONTROL_PROGRESS on the channel
* Start up the playback of the media
Instead, we sneak an answer on the channel right before starting playing media.
This is due to ARI's usage of control_streamfile. This function implicitly
answers the channel (and doesn't give ARI the option to stop it). The answering
of the channel here is probably unnecessary:
* app_voicemail, by far the biggest consumer of this function, always answers
the channels anyway
* control stream file (in res_agi) and ControlPlayback probably shouldn't be
implicitly answering the channel. Answering should not be tied directly to
playing back media.
As it turns out, the answering of the channel here is pretty old:
356042 twilson if (ast_channel_state(chan) != AST_STATE_UP) {
3087 anthm res = ast_answer(chan);
180259 tilghman }
(As in, ancient?)
Note that others ran into this problem and commented about it on various
mailing lists.
Review: https://reviewboard.asterisk.org/r/3907/
ASTERISK-24229 #close
Reported by: Matt Jordan
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Trivial patch to add new lines to several files missing them. This fixes
warnings when compiling with gcc 4.1.2 on CentOS 5.
ASTERISK-24245 #close
Reported by: Shaun Ruffell
patches:
0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417)
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This patch fixes gcc warnings that occur due to the type qualifier 'const'
being ignored on a return type of int.
ASTERISK-24246 #close
Reported by: Shaun Ruffell
patches:
0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417)
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On a SIP reinvite that changes media strams, the PJSIP channel driver was
flooding the log with "Asked to transmit frame type %s, while native
formats is %s" warnings.
* Fixes PJSIP not setting up translation paths when the formats change on
a reinvite. AFS-63 was effectively reintroduced because of the media
formats work. res_pjsip_sdp_rtp.c:set_caps()
* Improved the unexpected frame format WARNING message to include more
information.
* Added protective locking while altering formats on a channel. Reworked
set_format() to simplify and protect the formats under manipulation.
* Restored some code that got lost in the media_formats work.
(channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())
AFS-137 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3906/
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filename_completion_function() returns memory that was not allocated by
the MALLOC_DEBUG allocation tracker so the memory must be freed by
ast_std_free().
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This was causing the AMI show_subscriptions test in
the testsuite to fail since all subscriptions were being
seen as subscribers instead of notifiers.
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