Jonathan Rose [Fri, 19 Sep 2014 15:10:50 +0000 (15:10 +0000)]
Stasis_channels: Resolve unfinished Dials when doing masquerades
Masquerades into channels that are in the dialing state don't end their dial
and this goes against the model for things like CDRs and generating Dial end
manager actions and such.
ASTERISK-24237 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3990/
Kinsey Moore [Fri, 19 Sep 2014 12:30:39 +0000 (12:30 +0000)]
PJSIP: Prevent T38 framehook being put on wrong channel
This change gives framehooks a reverse-direction masquerade callback in
addition to chan_fixup_cb similar to the callback added to datastores
to handle the same situation. The new callback provides the same
parameters as the fixup callback, but is called on the new channel's
framehooks before moving framehooks from the old channel to the new
channel. This gives the framehooks an oppurtunity to decide whether
they should remain on the new channel or be removed.
This new callback is used to prevent the PJSIP T.38 framehook from
remaining on a masqueraded channel if the new channel is not also a
PJSIP channel. This was causing a crash when a local channel was
masqueraded into a PJSIP channel and the framehook was executed on the
local channel since the channel's tech private data was not structured
as expected.
George Joseph [Thu, 18 Sep 2014 19:21:56 +0000 (19:21 +0000)]
utils: Create ast_strsep function that ignores separators inside quotes
This function acts like strsep with three exceptions...
* The separator is a single character instead of a string.
* Separators inside quotes are treated literally instead of like separators.
* You can elect to have leading and trailing whitespace and quotes
stripped from the result and have '\' sequences unescaped.
Like strsep, ast_strsep maintains no internal state and you can call it
recursively using different separators on the same storage.
Also like strsep, for consistent results, consecutive separators are not
collapsed so you may get an empty string as a valid result.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3989/
Richard Mudgett [Thu, 18 Sep 2014 16:39:06 +0000 (16:39 +0000)]
astobj2.c/refcounter.py: Fix to deal with invalid object refs.
* Make astob2 REF_DEBUG output an invalid object line when an invalid ao2
object ref/unref is attempted. This is similar to the
constructor/destructor lines.
* Fixed refcounter.py to handle skewed objects that have
constructor/destructor states.
* Made refcounter.py highlight the invalid ao2 object refs by putting them
in their own section of the processed output file.
* Made refcounter.py highlight unreffing an object by more than one that
results in a negative ref count and the object being destroyed. The
abnormally destroyed object is reported in the invalid and finalized
object sections of the output.
Mark Michelson [Thu, 18 Sep 2014 15:40:47 +0000 (15:40 +0000)]
res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies.
res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE
arrives.
* It checks that there is a subscription handler for the Event
* It checks that there are body generators for the types in the Accept header
The problem is, there's nothing that ensures that these two things will
actually mesh with each other. For instance, Asterisk will accept a subscription
to MWI that accepts pidf+xml bodies. That doesn't make sense.
With this commit, we add some type information to the mix. Subscription
handlers state they generate data of type X, and body generators state
that they consume data of type X. This way, Asterisk doesn't end up in
some hilariously mismatched situation like the one in the previous paragraph.
George Joseph [Thu, 18 Sep 2014 15:01:11 +0000 (15:01 +0000)]
res_pjsip: ami: Fix error in AMI output when an endpoint has no transport
When no transport is associated to an endpoint, the AMI output for
PJSIPShowEndpoint indicates an error instead of silently ignoring the
missing transport.
This patch causes the error to appear only if a transport was specified
on the endpoint and the transport doesn't exist. It also fixes an issue
with counting the objects that were actually found.
ASTERISK-24161 #close
ASTERISK-24331 #close
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3998/
George Joseph [Thu, 18 Sep 2014 14:43:41 +0000 (14:43 +0000)]
config: bug: Fix SEGV in ast_category_insert when matching category isn't found
If you call ast_category_insert with a match category that doesn't exist, the
list traverse runs out of 'next' categories and you get a SEGV. This patch
adds check for the end-of-list condition and changes the signature to return
an int for success/failure indication instead of a void.
The only consumer of this function is manager and it was also changed to use
the return value.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3993/
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res_rtp_asterisk: Fix a myriad of TURN client issues.
1. The number of file descriptors an ioqueue instance can handle is fixed, so we
now spawn the required number to handle the load.
2. Our transport identifiers were exceeding the range supported by pjnath.
3. The TURN client did not set up client binding causing needless bandwidth usage.
4. The code no longer updates address information on each packet.
5. STUN traffic was getting looped back to Asterisk instead of going through the
TURN server.
6. Synchronization now ensures things are completely setup or destroyed.
7. Logging now reflects the target the TURN server is sending to/receiving from
on our behalf.
Kinsey Moore [Fri, 12 Sep 2014 18:19:48 +0000 (18:19 +0000)]
Blocked revisions 423010
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Bridging: Fix bouncing native bridge
This fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge
could cause a bouncing native bridge. In the case of the
dial_LS_options test, this was a remote RTP bridge which caused the
audio path to continually cycle between Asterisk and the remote
endpoints generating a large number of SIP messages and delaying the
test long enough to cause it to fail (checking timing was part of the
test). The root cause was that the code to decide whether to use native
bridging was expecting a time-remaining value of 0 to be the default
instead of the actual default value of -1. A value of 0 or negative
numbers could also be generated by preceding code in some
circumstances. Both issues are addressed in this patch.
ASTERISK-24211 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3987/
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Jonathan Rose [Fri, 12 Sep 2014 16:01:36 +0000 (16:01 +0000)]
Realtime: Fix a bug that caused realtime destroy command to crash
Also has could affect with anything that goes through ast_destroy_realtime.
If a CLI user used the command 'realtime destroy <family>' with only a single
column/value pair, Asterisk would crash when trying to create a variable list
from a NULL value.
Mark Michelson [Thu, 11 Sep 2014 22:16:03 +0000 (22:16 +0000)]
Remove undocumented default behavior of ast_play_and_record_full acceptdtmf.
ast_play_and_record_full() has a parameter called "acceptdtmf" that is a
string of acceptable DTMF digits that may be pressed by a caller to end
and accept the recording.
ARI uses this function in order to perform recording, and it provides
options for what is passed as acceptdtmf to ast_play_and_record_full().
By default, ARI passes an empty string, with the intention that no DTMF
can be used to end the recording.
The problem is that ast_play_and_record_full() attempts to be "helpful"
by setting "#" as the acceptdtmf if an empty string or NULL pointer
has been passed in. With ARI, this results in unexpected behavior
occurring if you have attempted to intercept "#" yourself in order
to perform some other manipulation of the live recording.
This change removes the "helpful" behavior by no longer accepting
"#" as a default acceptdtmf if none is specified by the caller of
ast_play_and_record_full(). This makes the ARI scenario work as
expected.
The other callers of ast_play_and_record_full() are app_voicemail
and app_minivm, and in both cases, they pass an explicit "#" to
ast_play_and_record_full() as acceptdtmf, so they are unaffected
by this change.
George Joseph [Wed, 10 Sep 2014 16:02:54 +0000 (16:02 +0000)]
config: bug: fix truncation of included config files on permissions error
ast_config_text_file_save() currently truncates include files as they
are processed. If a subsequent include file or the main config file has
a permissions error that prevents writing, earlier include files are left
truncated resulting in a frantic search for backups.
This patch causes ast_config_text_file_save to check for write access
on all files before it truncates any of them.
Will be applied 1.8 > trunk.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3986/
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Rusty Newton [Sun, 7 Sep 2014 00:09:54 +0000 (00:09 +0000)]
Sounds/BuildSystem: Modifications to include new releases and Japanese language.
Modifying Makefile and sounds.xml to include new core 1.4.26 and extra 1.4.15
sound prompt releases, plus the new Japanese core sound prompts contributed
by QLOOG.
ASTERISK-23324
Reported by: Kevin McCoy
Tested by: Rusty Newton
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Matthew Jordan [Sat, 6 Sep 2014 22:48:24 +0000 (22:48 +0000)]
main/cdr: Copy over location information during a fork
When a CDR is forked, a new CDR is created and appended to the CDR chain for
the Party A. The forked CDR starts life off as a clone of the last
non-finalized for the particular Party A. In the past, merely copying over
the snapshots for Party A/Party B would be sufficient. However, as the CDRs
now contain cached information from Party A - specifically application/data,
context, and extension - we need to copy that over during a fork as well.
Huzzah for unit tests catching this when the context/extension were derived
from a cached value on the CDR instead of on Party A.
Matthew Jordan [Sat, 6 Sep 2014 22:20:27 +0000 (22:20 +0000)]
main/rtp_engine: Format NTP timestamps as unsigned ints
On some systems, a timeval's tv_sec/tv_usec will be unsigned lont ints, as
opposed to long ints. When the RTP engine formats these as strings, it was
previously formatting them as signed integers, which can result in some
odd negative timestamp values (particularly on 32-bit systems). This patch
formats the values as unsigned long integers.
Matthew Jordan [Fri, 5 Sep 2014 22:02:23 +0000 (22:02 +0000)]
main/cdrs: Preserve context/extension when executing a Macro or GoSub
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:
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
Matthew Jordan [Fri, 5 Sep 2014 21:53:35 +0000 (21:53 +0000)]
main/cdr: Fix crash/memory consumption in CDRs in multi-party bridge scenarios
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.
Mark Michelson [Tue, 2 Sep 2014 18:16:55 +0000 (18:16 +0000)]
Resolve race condition where channels enter dialplan application before 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.
Matthew Jordan [Mon, 1 Sep 2014 14:07:04 +0000 (14:07 +0000)]
res_stasis: Don't play MoH to channels by default when added to holding bridges
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
George Joseph [Sat, 30 Aug 2014 17:28:04 +0000 (17:28 +0000)]
confbridge: Add Duration to ConfbridgeList event
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/
George Joseph [Sat, 30 Aug 2014 17:22:55 +0000 (17:22 +0000)]
manager: Make WaitEvent action respect eventfilters
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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Matthew Jordan [Fri, 29 Aug 2014 19:39:42 +0000 (19:39 +0000)]
doc: Add a manpage for the smsq utility
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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Matthew Jordan [Fri, 29 Aug 2014 19:32:49 +0000 (19:32 +0000)]
doc: Add a manpage for the aelparse utility
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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Matthew Jordan [Thu, 28 Aug 2014 21:53:51 +0000 (21:53 +0000)]
LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP
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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Michael L. Young [Thu, 28 Aug 2014 20:29:45 +0000 (20:29 +0000)]
chan_iax2: Fix Dynamic IAX2 Registrations After Temporary DNS Failure
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)
George Joseph [Wed, 27 Aug 2014 17:21:57 +0000 (17:21 +0000)]
confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events
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/
Kinsey Moore [Wed, 27 Aug 2014 15:14:39 +0000 (15:14 +0000)]
CallerID: Fix parsing of malformed callerid
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.
George Joseph [Tue, 26 Aug 2014 23:18:32 +0000 (23:18 +0000)]
confbridge: Make kick, mute and unmute handle channel targets consistently.
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/
Mark Michelson [Tue, 26 Aug 2014 22:08:39 +0000 (22:08 +0000)]
Fix race condition in the scheduler when deleting a running entry.
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.
Richard Mudgett [Mon, 25 Aug 2014 16:11:19 +0000 (16:11 +0000)]
res_musiconhold: Fix MOH restarting where it left off from the last hold.
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
Joshua Colp [Sun, 24 Aug 2014 19:34:50 +0000 (19:34 +0000)]
res_pjsip_transport_websocket: Attach the Websocket module on outgoing INVITEs.
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
Joshua Colp [Sun, 24 Aug 2014 19:18:51 +0000 (19:18 +0000)]
res_pjsip_transport_websocket: Fix a progressive memory growth.
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.
Joshua Colp [Sun, 24 Aug 2014 18:52:09 +0000 (18:52 +0000)]
res_pjsip_transport_websocket: Ensure secure Websocket clients can be called.
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.
Joshua Colp [Sun, 24 Aug 2014 17:20:29 +0000 (17:20 +0000)]
chan_sip: Use the server reflexive ICE candidate RTCP port as provided.
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.
Mark Michelson [Thu, 21 Aug 2014 21:42:08 +0000 (21:42 +0000)]
Switch from hostname to an IP address in the SDP origin line.
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.
Mark Michelson [Thu, 21 Aug 2014 21:35:21 +0000 (21:35 +0000)]
Ensure after-bridge behavior is correct when moving from Stasis to a 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.
Jonathan Rose [Thu, 21 Aug 2014 21:15:40 +0000 (21:15 +0000)]
res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set
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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Mark Michelson [Thu, 21 Aug 2014 21:14:20 +0000 (21:14 +0000)]
Improve consistency of party ID privacy usage.
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.
Matthew Jordan [Thu, 21 Aug 2014 17:33:56 +0000 (17:33 +0000)]
chan_sip: Don't use port derived from fromdomain if it isn't set
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.
Matthew Jordan [Thu, 21 Aug 2014 15:22:53 +0000 (15:22 +0000)]
ARI: Fix implicit answer when playback is initiated on unanswered channel
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.
Richard Mudgett [Wed, 20 Aug 2014 22:19:41 +0000 (22:19 +0000)]
cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled.
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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Mark Michelson [Wed, 20 Aug 2014 20:02:56 +0000 (20:02 +0000)]
Move evaluation of set_var options in pjsip to the end of channel initialization.
This allows for set_var to override certain defaults such as caller ID and codec
values. This also fixes a test suite regression. The "set_var" test suite test attempted
to use set_var to override caller ID, but a recent change caused that to no longer work.
Kinsey Moore [Wed, 20 Aug 2014 12:56:58 +0000 (12:56 +0000)]
Stasis: Add information to blind transfer event
When a blind transfer occurs that is forced to create a local channel
pair to satisfy the transfer request, information about the local
channel pair is not published. This adds a field to describe that
channel to the blind transfer message struct so that this information
is conveyed properly to consumers of the blind transfer message.
This also fixes a bug in which Stasis() was unable to properly identify
the channel that was replacing an existing Stasis-controlled channel
due to a blind transfer.
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3921/
Jonathan Rose [Tue, 19 Aug 2014 16:20:59 +0000 (16:20 +0000)]
ARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX
If /channels/{channelID}/continue is called on a channel that was originated
without a PBX (such as the ARI command POST channel with a stasis application
argument), the channel will not start dialplan execution. This patch will now
run the PBX out of the stasis execution if the channel doesn't currently have
an active PBX upon continuing.
Richard Mudgett [Tue, 19 Aug 2014 16:06:58 +0000 (16:06 +0000)]
chan_pjsip: Fix attended transfer connected line name update.
A calls B
B answers
B SIP attended transfers to C
C answers, B and C can see each other's connected line information
B completes the transfer
A has number but no name connected line information about C
while C has the full information about A
I examined the incoming and outgoing party id information handling of
chan_pjsip and found several issues:
* Fixed ast_sip_session_create_outgoing() not setting up the configured
endpoint id as the new channel's caller id. This is why party A got
default connected line information.
* Made update_initial_connected_line() use the channel's CALLERID(id)
information. The core, app_dial, or predial routine may have filled in or
changed the endpoint caller id information.
* Fixed chan_pjsip_new() not setting the full party id information
available on the caller id and ANI party id. This includes the configured
callerid_tag string and other party id fields.
* Fixed accessing channel party id information without the channel lock
held.
* Fixed using the effective connected line id without doing a deep copy
outside of holding the channel lock. Shallow copy string pointers can
become stale if the channel lock is not held.
* Made queue_connected_line_update() also update the channel's
CALLERID(id) information. Moving the channel to another bridge would need
the information there for the new bridge peer.
* Fixed off nominal memory leak in update_incoming_connected_line().
* Added pjsip.conf callerid_tag string to party id information from
enabled trust_inbound endpoint in caller_id_incoming_request().
George Joseph [Mon, 18 Aug 2014 20:17:09 +0000 (20:17 +0000)]
func_config: Change 'Not Found' message from ERROR to DEBUG
When you call the CONFIG dialplan function with the name of a variable that
doesn't exist in the target context you get an ERROR. This does nothing but
clutter up the logs with messages that may be perfectly acceptable. Just
because a variable wasn't in the context doesn't mean it's an error. Maybei
t's optional or just needs to be defaulted or ignored.
This patch changes the log level from ERROR to DEBUG. If a dialplan developer
wants to debug their dialplan they still canby setting the console debug level
as needed.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3919/
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Matthew Jordan [Sun, 17 Aug 2014 23:28:27 +0000 (23:28 +0000)]
apps/app_meetme: Fix crash when publishing MeetMe messages with no channel
The same function, meetme_stasis_generate_msg, handles creating and publishing
Stasis message both when there are channels in the MeetMe conference and when
there are no channels in the conference. When the performance improvement was
made to use cached snapshots, this created a situation where Asterisk would
crash: obtaining a cached snapshot is not NULL tolerant.
This patch restores the previous implementation, which used a NULL safe set
of routines to produce a blob containing the channel snapshot (if available)
and information about the MeetMe conference.
Matthew Jordan [Sun, 17 Aug 2014 23:08:57 +0000 (23:08 +0000)]
apps/app_dial: Fix Dial 'z' option
The 'z' option is supposed to disable the dial timeout in the case of a call
forward. Unfortunately, the wrong timeout timer was passed to the do_forward
function, resulting in the option not working.
Matthew Jordan [Sun, 17 Aug 2014 22:33:21 +0000 (22:33 +0000)]
configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc
Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is
executed with optimization. This "help" unfortunately results in re-definition
warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This
patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning.
Review: https://reviewboard.asterisk.org/r/3912/
ASTERISK-24032 #close
Reported by: Kilburn
Tested by: Kilburn, wdoekes
patches:
1.8.diff uploaded by cloos (License 5956)
10.diff uploaded by cloos (License 5956)
11.diff uploaded by cloos (License 5956)
12.diff uploaded by cloos (License 5956)
13.diff uploaded by cloos (License 5956)
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Jonathan Rose [Fri, 15 Aug 2014 16:56:33 +0000 (16:56 +0000)]
Bridging: Fix a behavioral change when checking if a channel is leaving a bridge
r420934 introduced some failures in the test suite. Upon investigating, it was
discovered that differences in the way we were evaluating whether a channel was in
the process of leaving a bridge were causing some reinvites not to occur (mostly
reinvites back to Asterisk when ending a call). This patch fixes that behavioral
change.
ASTERISK-24027 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3910/
Matthew Jordan [Fri, 15 Aug 2014 15:41:35 +0000 (15:41 +0000)]
app_voicemail/app: Remove test events that were duplicated by r421059
Moving the test event raised when a file is played back (which occurred in
r421059) broke the ever loving snot out of the voicemail tests. This caused
duplicate test events to get raised, as app_voicemail and main/app were raising
events prior to call ast_streamfile. The voicemail tests did not enjoy getting
multiple events.
Since raising the playback event in ast_streamfile is far more useful to the
vast majority of tests, this patch keeps the call there and simply removes the
extraneous calls that duplicated the event.
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Matthew Jordan [Thu, 14 Aug 2014 21:15:44 +0000 (21:15 +0000)]
res/res_hep_rtcp: Remove dependency on PJSIP
The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need
to be included, as the module does not using PJPROJECT any fashion.
Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as
a dependency, this also meant that res_hep_rtcp will fail to compile on a
system without PJPROJECT.
This patch removes the include.
Thanks to Damien Wedhorn for pointing this out in #asterisk-dev.
ASTERISK-24236 #close
Reported by: Damien Wedhorn, Matt Jordan
Tested by: Damien Wedhorn
Matthew Jordan [Thu, 14 Aug 2014 19:09:32 +0000 (19:09 +0000)]
cel: Make sure channels in extra fields include their unique IDs as well
CEL typically tracks a lot of information using the unique ID of the channel.
This is typically needed due to tying events together using the linked ID of
the various channels involved in a "call", which is derived from the channel ID
of the oldest channel involved in a bridge (or in the case of a Dial, the
parent channel).
Previously, we had updated the extra fields to include the involved channel
names, but forgot to put in the unique ID. This patch corrects that error.
Kinsey Moore [Wed, 13 Aug 2014 16:47:10 +0000 (16:47 +0000)]
PJSIP: Prevent crash no-URI contacts
This prevents a crash from occurring when a contact with no URI is used
for the creation of an outbound out-of-dialog request with no
associated endpoint.
Jonathan Rose [Wed, 13 Aug 2014 15:21:07 +0000 (15:21 +0000)]
Bridges: Fix feature interruption/unintended kick caused by external actions
If a manager or CLI user attached a mixmonitor to a call running a dynamic
bridge feature while in a bridge, the feature would be interrupted and the
channel would be forcibly kicked out of the bridge (usually ending the call
during a simple 1 to 1 call). This would also occur during any similar action
that could set the unbridge soft hangup flag, so the fix for this was to
remove unbridge from the soft hangup flags and make it a separate thing all
together.
Kinsey Moore [Mon, 11 Aug 2014 18:35:27 +0000 (18:35 +0000)]
Stasis: Allow internal channels directly into bridges
The patch to catch channels being shoehorned into Stasis() via external
mechanisms also happens to catch Announcer and Recorder channels
because they aren't known to be stasis-controlled channels in the usual
sense. This marks those channels as Stasis()-internal channels and
allows them directly into bridges.
Walter Doekes [Mon, 11 Aug 2014 10:37:41 +0000 (10:37 +0000)]
general: Fix memory Corruption in __ast_string_field_ptr_build_va.
If the space left in a stringfield is between 0 and
(alignof(ast_string_field_allocation)-1) adding new data would cause
memory corruption, because we would assume enough space (unsigned
underrun).
Thanks Arnd Schmitter for reporting and finding out the cause!
Kinsey Moore [Thu, 7 Aug 2014 20:23:30 +0000 (20:23 +0000)]
Stasis: Ensure transfer messages follow validation rules
This makes Stasis() event generation for transfer messages follow
validation rules. Currently, ast_json_null() is being used in place of
omitting a key entirely which falls afoul of these validation rules.
Kinsey Moore [Thu, 7 Aug 2014 15:19:53 +0000 (15:19 +0000)]
Stasis: Convey transfer information to applications
This fixes a class of issues where Stasis applications were not made
aware that their channels were being manipulated or replaced by
external entitiessuch as transfers, AMI commands, or dialplan
applications such as Bridge(). Inconsistent information such as
StasisEnd events with unknown channels as a result of masquerades has
also been corrected. To accomplish these fixes, several new fields
were added to blind and attended transfer messages as well as
StasisStart and BridgeAttendedTransfer Stasis events.
projects / thirdparty / asterisk.git / log
