The "Add qualify_timeout processing and eventing" patch introduced
an issue where contacts that had qualify_frequency set to 0 were
showing Unavailable instead Unknown. This patch checks for
qualify_frequency=0 and create an "Unknown" contact_status
with an RTT = 0.
Previously, the lack of contact_status implied Unknown but since
we're now changing endpoint state based on contact_status, I've
had to add new UNKNOWN status so that changes could trigger the
appropriate contact_status observers.
ASTERISK-24977: #close
Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7
Three fax related tests started failing as a result of changes made for
ASTERISK-24841:
tests/fax/pjsip/gateway_t38_g711
tests/fax/sip/gateway_mix1
tests/fax/sip/gateway_mix3
Historically, ast_channel_make_compatible() did nothing if the channels
were already "compatible" even if they had a sub-optimal translation path
already setup. With the changes from ASTERISK-24841 this is no longer
true in order to allow the best translation paths to always be picked. In
res_fax.c:fax_gateway_framehook() code manually setup the channels to go
through slin and then called ast_channel_make_compatible(). With the
previous version of ast_channel_make_compatible() this was always a
no-operation.
* Remove call to ast_channel_make_compatible() in fax_gateway_framehook()
that now undoes what was just setup when the framehook is attached.
* Fixed locking around saving the channel formats in
fax_gateway_framehook() to ensure that the formats that are saved are
consistent.
* Fix copy pasta errors in fax_gateway_framehook() that confuses read and
write when dealing with saved channel formats.
ASTERISK-24841
Reported by: Matt Jordan
Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d
A potential problem that can arise is the following:
* Bob's phone is programmed to automatically forward to Carol.
* Carol's phone is programmed to automatically forward to Bob.
* Alice calls Bob.
If left unchecked, this results in an endless loops of call forwards
that would eventually result in some sort of fiery crash.
Asterisk's method of solving this issue was to track which interfaces
had been dialed. If a destination were dialed a second time, then
the attempt to call that destination would fail since a loop was
detected.
The problem with this method is that call forwarding has evolved. Some
SIP phones allow for a user to manually forward an incoming call to an
ad-hoc destination. This can mean that:
* There are legitimate use cases where a device may be dialed multiple
times, or
* There can be human error when forwarding calls.
This change removes the old method of detecting forwarding loops in
favor of keeping a count of the number of destinations a channel has
dialed on a particular branch of a call. If the number exceeds the
set number of max forwards, then the call fails. This approach has
the following advantages over the old:
* It is much simpler.
* It can detect loops involving local channels.
* It is user configurable.
The only disadvantage it has is that in the case where there is a
legitimate forwarding loop present, it takes longer to detect it.
However, the forwarding loop is still properly detected and the
call is cleaned up as it should be.
Address review feedback on gerrit.
* Correct "mfgium" to "Digium"
* Decrement max forwards by one in the case where allocation of the
max forwards datastore is required.
* Remove irrelevant code change from pjsip_global_headers.c
ASTERISK-24958 #close
Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint. Only dynamic contact add/delete actions
update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.
This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...
1. A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message. The default is 3000ms. When the timer expires, the contact is
marked unavailable.
2. Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'. When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The
existing endpoint events are generated appropriately.
ASTERISK-24863 #close
Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>
This change makes the send_notify of the sub_tree
not happen when the sub_tree has been deleted due
to the notify call failing, which avoids a crash.
ASTERISK-24970 #close
Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf
This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
Since we currently have no control over pjproject transaction timeout, this
patch pulls the pjsip_endpt_send_request function out of pjproject and into
res_pjsip/endpt_send_transaction in order to implement that capability.
Now when the transaction is initiated, we also schedule our own pj_timer with
our own desired timeout.
If the transaction completes before either timeout, pjproject cancels its timer,
and calls our tsx callback where we cancel our timer and run the app callback.
If the pjproject timer times out first, pjproject calls our tsx callback where
we cancel our timer and run the app callback.
If our timer times out first, we terminate the transaction which causes
pjproject to cancel its timer and call our tsx callback where we run the app
callback.
Regardless of the scenario, pjproject is calling the tsx callback inside the
group_lock and there are checks in the callback to make sure it doesn't run
twice.
As part of this patch ast_sip_send_out_of_dialog_request was created to replace
its similarly named private function. It takes a new timeout argument in
milliseconds (<= 0 to disable the timeout).
ASTERISK-24863 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747
Currently when Asterisk starts initial qualifies of contacts are spread out
randomly between 0 and qualify_timeout to prevent network and system overload.
If a contact's qualify_frequency is 5 minutes however, that contact may be
unavailable to accept calls for the entire 5 minutes after startup. So while
staggering the initial qualifies is a good idea, basing the time on
qualify_timeout could leave contacts unavailable for too long.
This patch adds a new global parameter "max_initial_qualify_time" that sets the
maximum time for the initial qualifies. This way you could make sure that all
your contacts are initialy, randomly qualified within say 30 seconds but still
have the contact's ongoing qualifies at a 5 minute interval.
If max_initial_qualify_time is > 0, the formula is initial_interval =
min(max_initial_interval, qualify_timeout * random(). If not set,
qualify_timeout is used.
The default is "0" (disabled).
ASTERISK-24863 #close
Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
Tested-by: George Joseph <george.joseph@fairview5.com>
OPTIONAL_API has conditionals to define AST_OPTIONAL_API and
AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined.
Unfortunately this is inside the include protection block, so only the
first status of AST_API_MODULE is respected. For example res_monitor
is an optional API provider, but uses func_periodic_hook. This makes
func_periodic_hook non-optional to res_monitor.
ASTERISK-17608 #close
Reported by: Warren Selby
Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679
Add the .gitignore and .gitreview files to the asterisk repo.
NB: You can add local ignores to the .git/info/exclude file
without having to do a commit.
Common ignore patterns are in the top-level .gitignore file.
Subdirectory-specific ignore patterns are in their own .gitignore
files.
Change-Id: I4c8af3b8e3739957db545f7368ac53f38e99f696
Tested-by: George Joseph
Prior to this patch, the far_max_datagram value on the UDPTL structure would
remain -1 if the remote endpoint fails to provide the SDP media attribute
T38FaxMaxDatagram. This can result in the INVITE request being rejected. With
this patch, we will now properly initialize the value with either the default
value or with the value provided by pjsip.conf's t38_udptl_maxdatagram
parameter.
Review: https://reviewboard.asterisk.org/r/4589
ASTERISK-24928 #close
Reported by: Juergen Spies
Tested by: Juergen Spies
patches:
pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this patch, chan_pjsip/res_pjsip now sets the native formats to the
codecs negotiated by a call.
* The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native
formats to include all the negotiated audio codecs instead of only the
initial preferred audio codec and later the currently received audio
codec.
* The audio frame handling in channel.c:ast_read() is more streamlined and
will automatically adjust to changes in received frame formats. The new
policy is to remove translation and pass the new frame format to the
receiver except if the translation was to a signed linear format. A more
long winded version is commented in ast_read() along with some caveats.
* The audio frame handling in channel.c:ast_write() is more streamlined
and will automatically adjust any needed translation to changes in the
frame formats sent. Frame formats sent can change for many reasons such
as a recording is being played back or the bridged peer changed the format
it sends. Since it is a normal expectation that sent formats can change,
the codec mismatch warning message is demoted to a debug message.
* Removed the short circuit check in
channel.c:ast_channel_make_compatible_helper(). Two party bridges need to
make channels compatible with each other. However, transfers and moving
channels among bridges can result in otherwise compatible channels having
sub-optimal translation paths if the make compatible check is short
circuited. A result of forcing the reevaluation of channel compatibility
is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc
options take effect consistently now. It is unfortunate that these two
options are enabled by default and negate some of the benefits to the
changes in channel.c:ast_read() by forcing translation through signed
linear on a two party bridge.
* Improved the softmix bridge technology to better control the translation
of frames to the bridge. All of the incoming translation is now normally
handled by ast_read() instead of splitting any translation steps between
ast_read() and the slin factory. If any frame comes in with an unexpected
format then the translation path in ast_read() is updated for the next
frame and the slin factory handles the current frame translation.
This is the final patch in a series of patches aimed at improving
translation path choices. The other patches are on the following reviews:
https://reviewboard.asterisk.org/r/4600/https://reviewboard.asterisk.org/r/4605/
ASTERISK-24841 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4609/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for automatically detecting the type of DTMF that a
PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
the channel created for an endpoint will attempt to determine if RFC 4733
DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
for the channel will be set to inband.
Review: https://reviewboard.asterisk.org/r/4438
ASTERISK-24706 #close
Reported by: yaron nahum
patches:
yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While investigating other unload issues I realized that the load/unload process
for the config wizard was pretty ugly so I've refactored it as follows...
When the res_pjsip sorcery instance is created the config_wizard bumps it's own
module reference to prevent it from unloading while the sorcery instance is
still active. When res_pjsip unloads and it's sorcery instance is destroyed,
the config wizard unrefs itself which then allows itself to unload cleanly.
Since the config wizard now can't load after res_pjsip or unload before it
(which should have been the correct behavior all along), I was able to remove
the chunks of code in both load_module and unload_module that handled that case.
Ran the testsuite tests to insure there were no functional changes and REF_DEBUG
to insure that Asterisk was shutting down cleanly with no FRACKs or leaks.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4610/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the ChannelHold event was added, the 'musicclass' parameter was
erroneously removed. This caused the ChannelHold events to be rejected as
they failed model validation. This patch updates the Swagger schema such that
it now properly reflects the event that is being created.
Hooray for tests that catch things like this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_pjsip_phoneprov_provider was leaking references to phoneprov objects due to
a missing OBJ_NODATA in an ao2_callback in load_users(). Rather than adding the
OBJ_NODATA, I changed load_users to use a more straightforward ao2_iterator.
This plugged the leak but exposed an unload order issue between
res_pjsip_phoneprov_provider, res_phoneprov and res_pjsip.
res_pjsip_phoneprov_provider unloads first, then res_phoneprov, then res_pjsip.
Since res_pjsip_phoneprov_provider uses res_pjsip's sorcery instance, when it
unloads, it's objects are still in the sorcery instance. When res_pjsip
unloads, it destroys all its objects including res_pjsip_phoneprov_provider's.
The phoneprov destructor then attempts to unregister the extension from
res_phoneprov but because res_phoneprov is already cleaned up, its users
container is gone and we get a FRACK.
Simple solution, check for the NULL users container before attempting to remove
the entry. Duh.
Ran tests/res_phoneprov/res_phoneprov_provider. No leaks in
res_pjsip_phoneprov_provider and no FRACKs.
Reported-by: Corey Farrell
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4608/
ASTERISK-24935 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new session supplement that handles in-dialog OPTIONS
requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup
for the OPTIONS request would already have been done by the time the
session supplement receives the inbound request.
ASTERISK-24862 #close
Reported by: yaron nahum
patches:
res_pjsip_dlg_options.c submitted by yaron nahum (License 6676)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes autological comparison warnings in the following:
* chan_skinny: letohl may return a signed or unsigned value, depending on the
macro chosen
* func_curl: Provide a specific cast to CURLoption to prevent mismatch
* cel: Fix enum comparisons where the enum can never be negative
* enum: Fix comparison of return result of dn_expand, which returns a signed
int value
* event: Fix enum comparisons where the enum can never be negative
* indications: tone_data.freq1 and freq2 are unsigned, and hence can never be
negative
* presencestate: Use the actual enum value for INVALID state
* security_events: Fix enum comparisons where the enum can never be negative
* udptl: Don't bother to check if the return value from encode_length is less
than 0, as it returns an unsigned int
* translate: Since the parameters are unsigned int, don't bother checking
to see if they are negative. The cast to unsigned int would already blow
past the matrix bounds.
* res_pjsip_exten_state: Use a temporary value to cache the return of
ast_hint_presence_state
* res_stasis_playback: Fix enum comparisons where the enum can never be
negative
* res_stasis_recording: Add an enum value for the case where the recording
operation is in error; fix enum comparisons
* resource_bridges: Use enum value as opposed to -1
* resource_channels: Use enum value as opposed to -1
Review: https://reviewboard.asterisk.org/r/4533
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4533.patch submitted by dkdegroot (License 6600)
........
Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Without this patch, if a PJSIP endpoint with udptl enabled and authentication
set attempted to use sendFax, the FAX session would fail during setup. This
was because the invite issued in response to being auth challenged would cause
the PJSIP channel performing the FAX to receive a second T38 framehook and
this would cause frames to be consumed in an inappropriate manner.
ASTERISK-24933 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4577/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes several warnings pointed out by the clang compiler.
* chan_pjsip: Removed check for data->text, as it will always be non-NULL.
* app_minivm: Fixed evaluation of etemplate->locale, which will always
evaluate to 'true'. This patch changes the evaluation to use
ast_strlen_zero.
* app_queue:
- Fixed evaluation of qe->parent->monfmt, which always evaluates to
true. Instead, we just check to see if the dereferenced pointer
evaluates to true.
- Fixed evaluation of mem->state_interface, wrapping it with a call to
ast_strlen_zero.
* res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero.
Review: https://reviewboard.asterisk.org/r/4541
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4541.patch submitted by dkdegroot (License 6600)
........
Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If we receive a MESSAGE request that we cannot send a response
to, we should not send the incoming MESSAGE to the dialplan.
This commit should help the bouncing message_retrans test to
pass consistently.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some applications - such as SLA - a phone pressing hold should not behave
in the fashion that the Asterisk core would like it to. Instead, the hold
action has some application specific behaviour associated with it - such as
disconnecting the channel that initiated the hold; only playing MoH to channels
in the bridge if the channels are of a particular type, etc.
One way of accomplishing this is to use a framehook to intercept the
hold/unhold frames, raise an event, and eat the frame. Tasty. This patch
accomplishes that using a new dialplan function, HOLD_INTERCEPT.
In addition, some general cleanup of raising hold/unhold Stasis messages was
done, including removing some RAII_VAR usage.
Review: https://reviewboard.asterisk.org/r/4549/
ASTERISK-24922 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When setting the configuration option 'timers' equal to 'no' the bit flag was
not properly negated. This patch clears all associated flags and only sets the
specified one. pjsip will handle any necessary flag combinations. Also went
ahead and did similar for the '100rel' option.
ASTERISK-24910 #close
Reported by: Ray Crumrine
Review: https://reviewboard.asterisk.org/r/4582/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes a warning caught by clang, in which it detected that large
chunks of extconf were unused. Frankly, I wish we could pretend that all of
extconf was unused, but alas, that is not yet the case.
A few extraneous functions in the parking tests were removed as well, for
the same reason.
Review: https://reviewboard.asterisk.org/r/4553
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4553.patch submitted by dkdegroot (License 6600)
........
Merged revisions 434093 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_pjsip_phoneprov_provider is using ao2_callback with OBJ_MULTIPLE, then
ignoring the return. OBJ_NODATA flag was to prevent a reference leak, but
this caused the module to FRACK on unload. Revert change until this can
be investigated further.
ASTERISK-24935
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4578/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a change to align behavior with that of Asterisk 11 and previous versions.
In those versions, if a parked call were retrieved, and the call ended, the parked
call retriever would be hung up after the ParkedCall application ran. Prior to this
patch, in Asterisk 13, the same situation would result in the parked call retriever
falling through to additional priorities in the extension where the ParkedCall
application was called. With this patch, the behavior between Asterisk 11 and 13
aligns.
ASTERISK-24899 #close
Reported by Malcolm Davenport
Patches:
ASTERISK-24899.patch uploaded by Mark Michelson(license #5049)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Outbound SIP MESSAGEs had the potential to be sent out
of order from how they were specified in a set of
dialplan steps.
This change creates a serializer for sending outbound
MESSAGE requests on. This ensures that the MESSAGEs are
sent by Asterisk in the same order that they were sent
from the dialplan.
ASTERISK-24937 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/4579
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes some invalid enum conversion warnings caught by clang. In
particular:
* chan_sip: Several functions mixed usage of the st_refresher_param
enum and st_refresher enum. This patch corrects the functions to use the
right enum.
* chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state.
* strings: Fixed incorrect usage of AO2 flags with strings container.
* res_stasis: Change a return enumeration to stasis_app_user_event_res.
Review: https://reviewboard.asterisk.org/r/4535
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4535.patch submitted by dkdegroot (License 6600)
........
Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes clange compiler warnings for initializer overrides.
Specifically:
res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration
value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing
those enum values, we therefore initialize the value twice to two different
values, "tlsv1" and "default". This patch changes it to just initialize
the index in the array to "tlsv1".
Review: https://reviewboard.asterisk.org/r/4539/
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4539.patch submitted by dkdegroot (License 6600)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk had an issue where retransmissions of MESSAGE requests resulted in
Asterisk processing the retransmission as if it were a new MESSAGE request.
This patch fixes the issue by creating a transaction in PJSIP on the incoming
request. This way, if a retransmission arrives, the PJSIP transaction layer
will resend the response and Asterisk will not ever see the retransmission.
ASTERISK-24920 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/4532/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch updates the kqueue timing module to conform to current timer API.
This fixes issues with using the kqueue timing source on Asterisk 13 on
FreeBSD 10. These issues include:
- Remove support for kevent64(). The values used to support Asterisk timers
fit within 32bits and so can be handled on all platforms via kevent().
- Provide debug logging for, but do not track, unacked events. This matches
the behavior of all other timer implementations.
- Implement continuous mode by triggering and leaving active, a user event.
This ensures that the file descriptor for the timer returns immediately from
poll(), without placing the load of a high speed timer on the kernel.
- In kqueue_timer_get_max_rate(), don't overstate the capability of the timer.
On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer
type kqueue supports for timers.
- In kqueue_timer_get_event(), assume the caller woke up from poll() and just
return the mode the timer is currently in. This matches all other timer
implementations.
- Adjust the test code now that unacked events are not tracked.
Review: https://reviewboard.asterisk.org/r/4465/
ASTERISK-24857 #close
Reported by: scsiguy
Tested by: Ed Hynan
patches:
rb4465.patch submitted by scsiguy (License 6692)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes
caused by running PJSIP functions from non-PJSIP threads.
* Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing
crashes in some cases. In theory pj_shutdown() should take care of this.
* Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at
shutdown.
* Resolve leaked config global in res_pjsip_notify.
* Unregister pubsub pjsip service module.
* Implement cleanup for res_pjsip_session.
ASTERISK-24731 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4498/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* In res/res_sorcery_realtime.c: Broke long line.
* In main/bucket.c: Eliminated unnecessary NULL check as
ast_sorcery_unref() is NULL tolerant and set the global object to NULL
after unref in the system shutdown bucket_cleanup().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to a quirk in the configuration handling of res_xmpp, the 'autoregister'
setting was never actually processed. This was due to not properly copying
over the global settings to the client settings when applying the
configuration to the run-time object.
Review: https://reviewboard.asterisk.org/r/4496/
ASTERISK-14233
ASTERISK-24780 #close
Reported by: Simon Arlott
patches:
asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756)
........
Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Valgrind found some memory leaks associated with
ast_pjsip_rdata_get_endpoint(). The leaks would manifest when sending
responses to OPTIONS requests, processing MESSAGE requests, and
res_pjsip supplements implementing the incoming_request callback.
* Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in
res/res_pjsip.c:supplement_on_rx_request(),
res/res_pjsip/pjsip_options.c:send_options_response(),
res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and
res/res_pjsip_messaging.c:send_response().
* Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in
res/res_pjsip_nat.c:nat_on_rx_message().
* Fixed inconsistent but benign return value in
res/res_pjsip/pjsip_options.c:options_on_rx_request().
Review: https://reviewboard.asterisk.org/r/4511/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Valgrind found a memory leak and invalid access.
* Fix invalid access by sscanf() being fed a non-nul terminated string of
digits in res/res_pjsip_sdp_rtp.c:get_codecs().
* Fix memory leak in main/sorcery.c:sorcery_object_field_destructor().
* Fix potential NULL pointer dereference in
main/xmldoc.c:xmldoc_get_syntax_config_option().
Review: https://reviewboard.asterisk.org/r/4513/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Valgrind found some memory leaks associated with ast_sockaddr_resolve().
Most of the leaks had already been fixed by earlier memory leak hunt
patches. This patch performs an audit of ast_sockaddr_resolve() and found
one more.
* Fix ast_sockaddr_resolve() memory leak in
apps/app_externalivr.c:app_exec().
* Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs
parameter for safety so the pointer will never be uninitialized on return.
The same goes for res/res_pjsip_acl.c:extract_contact_addr().
* Made functions that call ast_sockaddr_resolve() with RAII_VAR()
controlling the addrs variable use ast_free instead of ast_free_ptr to
provide better MALLOC_DEBUG information.
Review: https://reviewboard.asterisk.org/r/4509/
........
Merged revisions 433056 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated some documentation stating that endpoint identifiers registered without
a name are place at the front of the lookup list. Also renamed register method
'ast_sip_register_endpoint_identifier_by_name' to
'ast_sip_register_endpoint_identifier_with_name'
ASTERISK-24840
Reported by: Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes previously reverted code that caused binary incompatibility
problems with some modules. And like the original patch it makes sure that
no matter what order the endpoint identifier modules were loaded, priority is
given based on the ones specified in the new global 'endpoint_identifier_order'
option.
ASTERISK-24840
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4489/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The res_pjsip modules were manually checking both name and number
presentation values when there is a function that determines the combined
presentation for a party ID struct. The function takes into account if
the name or number components are valid while the manual code rarely
checked if the data was even valid.
* Made use ast_party_id_presentation() rather than manually checking party
ID presentation values.
* Ensure that set_id_from_pai() and set_id_from_rpid() will not return
presentation values other than what is pulled out of the SIP headers. It
is best if the code doesn't assume that AST_PRES_ALLOWED and
AST_PRES_USER_NUMBER_UNSCREENED are zero.
* Fixed copy paste error in add_privacy_params() dealing with RPID
privacy.
* Pulled the id->number.valid test from add_privacy_header() and
add_privacy_params() up into the parent function add_id_headers() to skip
adding PAI/RPID headers earlier.
* Made update_connected_line_information() not send out connected line
updates if the connected line number is invalid. Lower level code would
not add the party ID information and thus the sent message would be
unnecessary.
* Eliminated RAII_VAR usage in send_direct_media_request().
Review: https://reviewboard.asterisk.org/r/4472/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to a break in binary compatibility with some other modules these changes
are being reverted until the issue can be resolved.
ASTERISK-24840
Reported by: Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a type=global section is not defined in pjsip.conf the global
defaults are not applied. As a result the mandatory Max-Forwards header
is not added to SIP messages for res_pjsip/chan_pjsip.
The handling of pjsip.conf type=global objects has several problems:
1) If the global object is missing the defaults are not applied.
2) If the global object is missing the default_outbound_endpoint's default
value is not returned by ast_sip_global_default_outbound_endpoint().
3) Defines are needed so default values only need to be changed in one
place.
* Added a sorcery instance observer callback to check if there were any
type=global sections loaded. If there were more than one then issue an
error message. If there were none then apply the global defaults.
* Fixed ast_sip_global_default_outbound_endpoint() to return the
documented default when no type=global object is defined.
* Made defines for the global default values.
* Increased the default_useragent[] size because SVN version strings can
get lengthy and 128 characters may not be enough.
* Fixed an off-nominal code path ref leak in global_alloc() if the string
fields fail to initialize.
* Eliminated RAII_VAR in get_global_cfg() and
ast_sip_global_default_outbound_endpoint().
ASTERISK-24807 #close
Reported by: Anatoli
Review: https://reviewboard.asterisk.org/r/4467/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When escaping backslashes with MySQL, the proper way to escape the characters
in a LIKE clause is to escape the '\' four times, i.e., '\\\\'. To quote the
MySQL manual:
"Because MySQL uses C escape syntax in strings (for example, “\n” to represent
a newline character), you must double any “\” that you use in LIKE strings.
For example, to search for “\n”, specify it as “\\n”. To search for “\”,
specify it as “\\\\”; this is because the backslashes are stripped once by the
parser and again when the pattern match is made, leaving a single backslash to
be matched against."
ASTERISK-24808 #close
Reported by: Javier Acosta
patches:
res_config_odbc.diff uploaded by Javier Acosta (License 6690)
........
Merged revisions 432720 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A race condition happened between initiating a transfer and requesting
that a dialog termination be delayed. Occasionally, the transferrer
channels would exit the bridge and hangup before the dialog termination
delay was requested.
* Made request dialog termination delay before initiating the transfer
action. If the transfer fails then cancel the delayed dialog termination
request.
ASTERISK-24755 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4460/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It's possible to have a scenario that will create a conflict between endpoint
identifiers. For instance an incoming call could be identified by two different
endpoint identifiers and the one chosen depended upon which identifier module
loaded first. This of course causes problems when, for example, the incoming
call is expected to be identified by username, but instead is identified by ip.
This patch adds a new 'global' option to res_pjsip called
'endpoint_identifier_order'. It is a comma separated list of endpoint
identifier names that specifies the order by which identifiers are processed
and checked.
ASTERISK-24840 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4455/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As pjproject is now used as a shared library a different define,
HAVE_PJPROJECT, is used to specify if pjproject is present.
ASTERISK-24830 #close
Reported by: Stefan Engström
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Made safely get the TRANSFER_CONTEXT channel value while the channel is
locked in refer_incoming_attended_request() and
refer_incoming_blind_request(). The pointer returned by
pbx_builtin_getvar_helper() is only valid while the channel is locked.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Unfortunately, while initial testing with ConfBridge did not reproduce the
audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing
did show that bridge_softmix and/or ConfBridge has a severe problem bridging
two or more participants at different sampling rates. Sometimes, it even picks
odd sampling rates that cause hideous audio problems.
This patch backs out the offending portion of the code until the issues in
the affected bridging modules can be more properly analyzed.
ASTERISK-24841
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sending the following ARI commands caused Asterisk to crash if the JSON
body 'variables' object passes values of types other than strings.
POST /ari/channels
POST /ari/channels/{channelid}
PUT /ari/endpoints/sendMessage
PUT /ari/endpoints/{tech}/{resource}/sendMessage
* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),
ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and
ast_ari_endpoints_send_message_to_endpoint().
ASTERISK-24751 #close
Reported by: jeffrey putnam
Review: https://reviewboard.asterisk.org/r/4447/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some WebSocket applications, like [chan_respoke][], require a larger
frame size than the default 8k; this patch bumps the default to 16k.
This patch also fixes some problems exacerbated by large frames.
The sanity counter was decremented on every fread attempt in
ws_safe_read(), regardless of whether data was read from the socket or
not. For large frames, this could result in loss of sanity prior to
reading the entire frame. (16k frame / 1448 bytes per segment = 12
segments).
This patch changes the sanity counter so that it only decrements when
fread() doesn't read any bytes. This more closely matches the original
intention of ws_safe_read(), given that the error message is
"Websocket seems unresponsive".
This patch also properly logs EOF conditions, so disconnects are no
longer confused with unresponsive connections.
[chan_respoke]: https://github.com/respoke/chan_respoke
Review: https://reviewboard.asterisk.org/r/4431/
........
Merged revisions 432236 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses the following problems:
* ari/resource_channels: In ARI, we currently create a format capability
structure of SLIN and apply it to the new channel being created. This was
originally done when the PBX core was used to create the channel, as there
was a condition where a newly created channel could be created without any
formats. Unfortunately, now that the Dial API is being used, this has two
drawbacks:
(a) SLIN, while it will ensure audio will flows, can cause a lot of
needless transcodings to occur, particularly when a Local channel is
created to the dialplan. When no format capabilities are available, the
Dial API handles this better by handing all audio formats to the requsted
channels. As such, we defer to that API to provide the format
capabilities.
(b) If a channel (requester) is causing this channel to be created, we
currently don't use its format capabilities as we are passing in our own.
However, the Dial API will use the requester channel's formats if none
are passed into it, and the requester channel exists and has format
capabilities. This is the "best" scenario, as it is the most likely to
create a media path that minimizes transcoding.
Fixing this simply entails removing the providing of the format capabilities
structure to the Dial API.
* chan_pjsip: Rather than blindly picking the first format in the format
capability structure - which actually *can* be a video or text format - we
select an audio format, and only pick the first format if that fails. That
minimizes the weird scenario where we attempt to transcode between video/audio.
* res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure.
Since ast_request already limits us down to one format capability once the
format capabilities are passed along, there's no reason to squelch it here.
* channel: Fixed a comment. The reason we have to minimize our requested
format capabilities down to a single format is due to Asterisk's inability
to convey the format to be used back "up" a channel chain. Consider the
following:
PJSIP/A => L;1 <=> L;2 => PJSIP/B
g,u,a g,u,a g,u,a u
That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials
PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local
channel has inherited those format capabilities down the line; PJSIP/B
supports only ulaw. According to these format capabilities, ulaw is
acceptable and should be selected across all the channels, and no
transcoding should occur. However, there is no way to convey this: when L;2
and PJSIP/B are put into a bridge, we will select ulaw, but that is not
conveyed to PJSIP/A and L;1. Thus, we end up with:
PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B
g g X u u
Which causes g722 to be written to PJSIP/B.
Even if we can convey the 'ulaw' choice back up the chain (which through
some severe hacking in Local channels was accomplished), such that the chain
looks like:
PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B
u u u u
We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back
with only 'ulaw'. This results in all the channel structures being set up
correctly, but PJSIP/A *still* sending g722 and causing the chain to fall
apart.
There's a lot of difficulty just in setting this up, as there are numerous
race conditions in the act of bridging, and no clean mechanism to pass the
selected format backwards down an established channel chain. As such, the
best that can be done at this point in time is clarifying the comment.
Review: https://reviewboard.asterisk.org/r/4434/
ASTERISK-24812 #close
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change makes it so that if a channel variable is requested and it does not exist
a 404 response will be returned instead of an allocation failed response. This makes
it easier to debug and figure out what is going on for a user.
ASTERISK-24677 #close
Reported by: Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some implementations don't pay attention to the expires for individual contacts.
In this case they may consider the lack of an Expires header in the 200 OK as
unregistered. This change makes it so if an Expires header is present in the REGISTER
we will add one in the 200 OK.
ASTERISK-24785 #close
Reported by: Ross Beer
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matt Hoskins reported that res_pjsip_publish_asterisk wouldn't pull config from
realtime. Turns out it was just missing a call ast_sorcery_apply_config().
res_pjsip_acl was missing it as well, so I added it. The other pjsip modules
looked OK.
ASTERISK-24811 #close
Reported-by: Matt Hoskins
Tested-by: George Joseph
Tested-by: Matt Hoskins
patches:
res_pjsip_publish_asterisk.c.patch submitted by Matt Hoskins (license 6688)
Review: https://reviewboard.asterisk.org/r/4433/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed hangup handling of the session->channel after answer if the
ast_channel_move() or ast_bridge_impart() fails. We are still the thread
controlling the session->channel so we need to call ast_hangup() to kill
the channel.
* Fixed debug messages in refer_incoming_invite_request() referencing
incorrect channnels on success. Code comments now say why the
session->channel cannot be used.
Review: https://reviewboard.asterisk.org/r/4422/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Analyzing a one-off crash on a busy system showed that processing a REFER
request had a NULL session channel pointer. The only way I can think of
that could cause this is if an outgoing BYE transaction overlapped the
incoming REFER transaction in a collision. Asterisk sends a BYE while the
phone sends a REFER to complete an attended transfer.
* Made check the session channel pointer before processing an incoming
REFER request in res_pjsip_refer.
* Fixed similar crash potential for res_pjsip supplement incoming request
processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE,
res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER
messages.
* Made res_pjsip_messaging respond to a message body too large with a 413
instead of ignoring it.
ASTERISK-24700 #close
Reported by: Zane Conkle
Review: https://reviewboard.asterisk.org/r/4417/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When RTCP debugging was enabled, an RTCP report without a report block would
cause a crash. This was due to the verbose output not checking to see if the
report_block pointer was NULl before dereferencing it.
This patch adds the necessary check to prevent printing any verbose output
if the far side hasn't provided us the information they should have.
ASTERISK-24791 #close
Reported by: JoshE
Tested by: JoshE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The res_sorcery_config module currently uses a fixed bucket
size of 53. This means that depending on the number of objects
you either end up with excess buckets or a lot of collisions.
Due to the way that res_sorcery_config is implemented it's actually
possible to make the bucket size dynamic based on the number of
objects. This is due to the fact that each loading of the config file
produces a new container and does not modify the existing one.
This change uses the number of expected objects and finds a prime
number near it. In practice depending on the number of objects this
can speed up lookups anywhere from 2X to 15X. This change also removes
the lock from the container as it is not needed.
Review: https://reviewboard.asterisk.org/r/4423/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When debugging things it can be useful to know absolutely what
version of pjproject res_pjsip is running against. This change
adds a "pjsip show version" CLI command which can be used to
query for this.
ASTERISK-24685 #close
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/4424/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During some refactoring the way private information for timers
was stored was changed. As a result of this the action which normally
removed the timer upon closure in res_timing_pthread was also removed
causing the timer to remain after it should using up resources.
This change ensures that the timer is removed upon closure.
ASTERISK-24768 #close
Reported by: Matthias Urlichs
patches:
timer.patch submitted by Matthias Urlichs (license 5508)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A multi-asterisk box setup with direct media enabled would occasionally
crash when two re-INVITE collisions on a call leg happen in a row.
The re-INVITE logic only had one timer struct to defer the re-INVITE.
When the second collision happens the timer struct is overwritten and put
into the timer heap again. Resources for the first timer are leaked and
the heap has two positions occupied by the same timer struct. Now the
heap ordering is potentially corrupted, the timer will fire twice, and any
resources allocated for the second timer will be released twice.
* The solution is to put the collided re-INVITE into the delayed requests
queue with all the other delayed requests and cherry pick the next request
that can come off the queue when an event happens.
* Changed to put delayed BYE requests at the head of the delayed queue.
There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE
has been requested.
* Made the start of a BYE request flush the delayed requests queue to
prevent a delayed request from overlapping the BYE transaction. I saw a
few cases where a delayed re-INVITE got started after the BYE transaction
started.
* Changed the delayed_request struct to use an enum instead of a string
for the request method. Cherry picking the queue is easier with an enum
than string comparisons and the compiler can warn if a switch statement
does not cover all defined enum values.
* Improved the debug output to give more information. It helps to know
which channel is involved with an endpoint. Trunks can have many channels
associated with the endpoint at the same time.
ASTERISK-24727 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4414/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new feature to ARI to redirect a channel to another server,
and fixes a few bugs in PJSIP's handling of the Transfer dialplan
application/ARI redirect capability.
*New Feature*
A new operation has been added to the ARI channels resource, redirect. With
this, a channel in a Stasis application can be redirected to another endpoint
of the same underlying channel technology.
*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP
stack:
(1) The existing .transfer channel callback had the limitation that it could
only transfer channels to a SIP URI, i.e., you had to pass
'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
still supported, it is somewhat unintuitive - particularly in a world full
of endpoints. As such, we now also support specifying the PJSIP endpoint to
transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
updating its Contact header. Alas, that resulted in the forwarding
destination set by the dialplan application/ARI resource/whatever being
rewritten with very incorrect information. Hence, we now don't bother
updating an outgoing response if it is a 302. Since this took a looong time
to find, some additional debug statements have been added to those modules
that update the Contact headers.
Review: https://reviewboard.asterisk.org/r/4316/
ASTERISK-24015 #close
Reported by: Private Name
ASTERISK-24703 #close
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There have been a couple of times where a crash occurred in the dtls_handler
section of the code for res_pjsip. Unfortunately, in working this issue the
problem was unable to be reproduced. After looking at the backtraces and
through the code the current best guess as to why this happened might be due
to a reentrance problem and the strtok function. So, the current fix is to
convert the strtok function into the reentrant version of the function,
strtok_r.
ASTERISK-24741 #close
Reported by: Zane Conkle
Review: https://reviewboard.asterisk.org/r/4409/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are three CLI commands to stop and restart Asterisk each.
1) core stop/restart now - Hangup all calls and stop or restart Asterisk.
New channels are prevented while the shutdown request is pending.
2) core stop/restart gracefully - Stop or restart Asterisk when there are
no calls remaining in the system. New channels are prevented while the
shutdown request is pending.
3) core stop/restart when convenient - Stop or restart Asterisk when there
are no calls in the system. New calls are not prevented while the
shutdown request is pending.
ARI has made stopping/restarting Asterisk more problematic. While a
shutdown request is pending it is desirable to continue to process ARI
HTTP requests for current calls. To handle the current calls while a
shutdown request is pending, a new committed to shutdown phase is needed
so ARI applications can deal with the calls until the system is fully
committed to shutdown.
* Added a new shutdown committed phase so ARI applications can deal with
calls until the final committed to shutdown phase is reached.
* Made refuse new HTTP requests when the system has reached the final
system shutdown phase. Starting anything while the system is actively
releasing resources and unloading modules is not a good thing.
* Split the bridging framework shutdown to not cleanup the global bridging
containers when shutting down in a hurry. This is similar to how other
modules prevent crashes on rapid system shutdown.
* Moved ast_begin_shutdown(), ast_cancel_shutdown(), and
ast_shutting_down(). You should not have to include channel.h just to
access these system functions.
ASTERISK-24752 #close
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/4399/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Looking at the Super Awesome Company sample reminded me that creating hints is
just plain gruntwork. So you can now have the pjsip conifg wizard auto-create
them for you.
Specifying 'hint_exten' in the wizard will create
'exten => <hint_exten>,hint/PJSIP/<wizard_id>'
in whatever is specified for 'hint_context'.
Specifying 'hint_application' in the wizard will create
'exten => <hint_exten>,1,<hint_application>'
in whatever is specified for 'hint_context'.
The default for 'hint_context' is the endpoint's context.
There's no default for 'hint_application'. If not specified, no app is added.
There's no default for 'hint_exten'. If not specified, neither the hint itself
nor the application will be created.
Some may think this is the slippery slope to users.conf but hints are a basic
necessity for phones unlike voicemail, manager, etc that users.conf creates.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4383/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
One of the canonical reasons for hanging up a channel is because the far end
failed to answer - or because someone else answered, and we want to get rid of
this channel. This patch adds the missing value to the 'reason' query parameter
for the DELETE /channels operation.
Review: https://reviewboard.asterisk.org/r/4400
ASTERISK-24745 #close
Reported by: Ben Merrills
patches:
add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies the ast_odbc_find_table function such that it only performs
a lookup of the requested table if the table is not already known. Prior to
this patch, a queries would be executed against the database even if the table
was already known and cached.
Review: https://reviewboard.asterisk.org/r/4405/
ASTERISK-24742 #close
Reported by: ibercom
patches:
patch.diff uploaded by ibercom (License 6599)
........
Merged revisions 431617 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an SDP is created for an outgoing request/response, the ICE candidates
obtained from the RTP instance are currently leaked. This causes the ao2
container that holds the candidates to never properly be reclaimed when the
RTP instance is destroyed.
This patch properly decrements the ICE candidates' container if it is
successfully obtained.
ASTERISK-24769 #close
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk attempts to send SIP outbound publish information and no response
is ever received (no 200 okay, 412, 423) the system eventually crashes. A
response is never received because the system Asterisk is attempting to send
publish information to is not available. The underlying pjsip framework attempts
to send publish information. After several attempts it calls back into the
Asterisk outbound publish code. At this point if the "client->queue" is empty
Asterisk attempts to schedule a refresh which utilizes "rdata" and since no
response was received the given "rdata" struture is NULL. Attempting to
dereference a NULL object of course results in a crash.
The fix here removes the dependency on rdata for schedule_publish_refresh.
Instead param->expiration is now passed to it as this is set to -1 if no
response is received. Also added a notification when no response is received.
ASTERISK-24635 #close
Reported by: Marco Paland
Review: https://reviewboard.asterisk.org/r/4384/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When swapping a Local channel in place of one already
in a bridge (to complete a bridge attended transfer),
the channel that was swapped out can actually be hung
up before the stasis bridge push callback executes on
the independant transfer thread. This results in the
stasis app loop dropping out and removing the control
that has the the app name which the local replacement
channel needs so it can re-enter stasis.
To avoid this race condition a new push_peek callback
has been added, and called from the ast_bridge_impart
thread before it launches the independant thread that
will complete the transfer. Now the stasis push_peek
callback can copy the stasis app name before the swap
channel can hang up.
ASTERISK-24649
Review: https://reviewboard.asterisk.org/r/4382/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
scenarios when we are required to use SIPS URIs in Contact
headers. Asterisk's non-compliance with this could actually
cause calls to get dropped when communicating with clients
that are strict about checking the Contact header.
Both of the SIP stacks in Asterisk suffered from this issue.
This changeset corrects the behavior in res_pjsip/chan_pjsip.c
Review: https://reviewboard.asterisk.org/r/4345
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to an inversion error, setting 100rel=no would not actually
change the current value of the setting (which defaulted to "yes").
With this fix, the inversion is corrected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent security fix for OpenSSL broke DTLS negotiation for many
applications. This was caused by read ahead not being enabled when it
should be. While a commit has gone into OpenSSL to force read ahead
on for DTLS it may take some time for a release to be made and the
change to be present in distributions (if at all). As enabling read
ahead is a simple one line change this commit does that and fixes
the issue.
ASTERISK-24711 #close
Reported by: Jared Biel
........
Merged revisions 431384 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r419044, we changed how formats were handled, but the return value
of the format_parse_sdp_fmtp functions in res_format_attr_opus and
res_format_attr_silk were not updated, causing calls to fail. Ran
into this when getting codec_opus working with Asterisk 13.
Once the return value was corrected, we were crashing in opus_getjoint
because of NULL format attributes. I've fixed this as well in this
patch.
Review: https://reviewboard.asterisk.org/r/4371/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Performing a CLI "module reload" command when there are new pjsip.conf
registration objects defined frequently failed to load them correctly.
What happens is a race condition between res_pjsip pushing its reload into
an asynchronous task processor task and the thread that does the rest of
the reloads when it gets to reloading the res_pjsip_outbound_registration
module. A similar race condition happens between a reload and the CLI/AMI
show registrations commands. The reload updates the current_states
container and the CLI/AMI commands call get_registrations() which builds a
new current_states container.
* Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous()
instead of ast_sip_push_task() to eliminate two threads processing config
reloads at the same time.
* Made get_registrations() not replace the global current_states container
so the CLI/AMI show registrations command cannot interfere with reloading.
You could never add/remove objects in the container without the
possibility of the container being replaced out from under you by
get_registrations().
* Added a registration loaded sorcery instance observer to purge any dead
registration objects since get_registrations() cannot do this job anymore.
The struct ast_sorcery_instance_observer callbacks must be used because
the callback happens inline with the load process. The struct
ast_sorcery_observer callbacks are pushed to a different thread.
* Added some global current_states NULL pointer checks in case the
container disappears because of unload_module().
* Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded
callbacks guaranteed to be called before any struct
ast_sorcery_observer.loaded callbacks will be called.
* Moved the check for non-reloadable objects to before the sorcery
instance loading callbacks happen to short circuit unnecessary work.
Previously with non-reloadable objects, the sorcery instance
loading/loaded callbacks would always happen, the individual wizard
loading/loaded would be prevented, and the non-reloadable type logging
message would be logged for each associated wizard.
ASTERISK-24729 #close
Review: https://reviewboard.asterisk.org/r/4381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to the original patch causing memory corruptions it was removed until the
problem could be resolved. This patch is the original patch plus some added
locking around stasis router subcription that was needed to avoid the memory
corruption.
Description of the original problem and patch (still applicable):
The res_pjsip module was previously unloadable. With this patch it can now
be unloaded.
This patch is based off the original patch on the issue (listed below) by Corey
Farrell with a few modifications. Namely, removed a few changes not required to
make the module unloadable and also fixed a bug that would cause asterisk to
crash on unloading.
This patch is the first step (should hopefully be followed by another/others at
some point) in allowing res_pjsip and the modules that depend on it to be
unloadable. At this time, res_pjsip and some of the modules that depend on
res_pjsip cannot be unloaded without causing problems of some sort.
The goal of this patch is to get res_pjsip and only res_pjsip to be able to
unload successfully and/or shutdown without incident (crashes, leaks, etc...).
Other dependent modules may still cause problems on unload.
Basically made sure, with the patch applied, that res_pjsip (with no other
dependent modules loaded) could be succesfully unloaded and Asterisk could
shutdown without any leaks or crashes that pertained directly to res_pjsip.
ASTERISK-24485 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4363/
patches:
pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change fixes two issues:
1. During a swap operation bridging added the new channel before having the swap channel
leave. This was not handled in bridge_native_rtp and could result in a channel not getting
reinvited back to Asterisk. After this change the swap channel will leave first and the
new channel will then join.
2. If a re-invite was received after a session had been established any upstream elements
(such as bridge_native_rtp) were not notified that they may want to re-evaluate things.
After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs
and upstream can react.
AST-1524 #close
Review: https://reviewboard.asterisk.org/r/4378/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch improves the documentation of ARI on the wiki. Specifically, it
addresses the following:
* Allowed values and allowed ranges weren't documented. This was particularly
frustrating, as Asterisk would reject query parameters with disallowed values
- but we didn't tell anyone what the allowed values were.
* The /play/id operation on /channels and /bridges failed to document all of
the added media resource types.
* Documentation for creating a channel into a Stasis application failed to
note when it occurred, and that creating a channel into Stasis conflicts with
creating a channel into the dialplan.
* Some other minor tweaks in the mustache templates, including italicizing the
parameter type, putting the default value on its own sub-bullet, and some
other nicities.
Review: https://reviewboard.asterisk.org/r/4351
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is currently a race condition when unloading the res_parking
module. Depending on the will of the universe the subscription
invocation may occur AFTER the module is unloaded. This is because
the module does NOT use stasis_unsubscribe_and_join when terminating
the subscription. It merely uses stasis_unsubscribe.
This change makes it use stasis_unsubscribe_and_join which is documented
for usage in this exact scenario.
AST-1520 #close
Review: https://reviewboard.asterisk.org/r/4375/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses compilation errors on OS X. It's been a while, so
there's quite a few things.
* Fixed __attribute__ decls in route.h to be portable.
* Fixed htonll and ntohll to work when they are defined as macros.
* Replaced sem_t usage with our ast_sem wrapper.
* Added ast_sem_timedwait to our ast_sem wrapper.
* Fixed some GCC 4.9 warnings using sig*set() functions.
* Fixed some format strings for portability.
* Fixed compilation issues with res_timing_kqueue (although tests still fail
on OS X).
* Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue
on OS X).
ASTERISK-24539 #close
Reported by: George Joseph
ASTERISK-24544 #close
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/4327/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After a bridge transfer completes where a local replacement
channel is used, a stasis transfer message with the details
of the transfer is sent. This is processed by stasis which
then sets the stasis app name and replaced channel snapshot
on the replacement channel.
However, since a separate thread was already started to run
stasis on the new replacement channel, a race was on to see
if the message processing would be completed before the app
name was needed, otherwise the channel would be hung up.
This change moves the calls used to set the stasis app name
and the replace snapshot to the bridge_stasis_push function
callback from the bridge transfer logic, allowing the steps
to be completed earlier and more deterministically, and the
race elimianted.
NOTE: the swap channel parameter to bridge_stasis_push (and
thus all bridge push callbacks) must always be present when
performing a swap with another channel.
ASTERISK-24649 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4341/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this changeset, posting to the: /ari/bridges/{bridgeId} endpoint without specifying a value for the [name] query parameter, would crash Asterisk if the bridge you are attempting to create (or update) had the same ID as an existing bridge. The internal mechanism of the POST operation interpreted a null value for name, thus resulting in an error condition that crashed Asterisk.
ASTERISK-24560 #close
Reported By: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/4349/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Calling ast_channel_bridge_peer() cannot be done while holding any channel
locks. The reported issue hit the deadlock in chan_iax2, but an audit of
the ast_channel_bridge_peer() calls found three more locations where the
same deadlock can occur.
* Made CHANNEL(peer), res_fax, and the SNMP agent not call
ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I
had to rework the logic to not hold the channel lock.
* Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done
for legacy reasons that no longer apply.
* Removed the iax.conf forcejitterbuffer option. It is now always enabled
when the jitterbuffer option is enabled. If you put a jitter buffer on a
channel it will be on the channel.
ASTERISK-24600 #close
Reported by: Jeff Collell
Review: https://reviewboard.asterisk.org/r/4342/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The first thing this patch fixes is UAS dialogs. Previously if a transport was
configured on an endpoint and an inbound session was created there was no guarantee
that requests sent on the dialog would use the correct transport and address
information. This has now been fixed so an explicitly configured transport
is taken into account.
The second thing this patch fixes is res_pjsip_multihomed. The res_pjsip_multihomed
module attempts to determine what transport a message should go out on and what
addressing information should go into the message itself. In a scenario where
multiple transports exist bound to the same IP address but a different port the
code would incorrectly alter the transport and change the message to the wrong
transport. This change makes the res_pjsip_multihomed module smarter so it will
only change the transport and address information in the message when it is
possible and makes sense.
ASTERISK-24615 #close
Reported by: David Justl
Review: https://reviewboard.asterisk.org/r/4331/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Different clients react differently to being told that a blind transfer
has failed. Some will simply send a BYE and be done with it. Others will
attempt to reinvite themselves back onto the call.
In the latter case, we were creating a new channel and then leaving it to
sit forever doing nothing. With this code change, that new channel will
not be created and the dialog with the transferring channel will be cleaned
up properly.
ASTERISK-24624 #close
Reported by Zane Conkle
Review: https://reviewboard.asterisk.org/r/4339
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When FAX was developed, apparently the faxregistry.container used to be a
linked list that was converted to an ao2 container. Some of the
replacement ao2 container operations still had explicit lock/unlocks
around them.
Three off nominal code paths in res_fax.c and res_fax_spandsp.c unlock the
channel even though the routine did not lock the channel and other code
paths in the routine do not unlock the channel.
Review: https://reviewboard.asterisk.org/r/4340/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to the split of outbound registration state from configuration it is possible during
a reload for a "pjsip show registrations" CLI command to be executed which gets an older
snapshot of the configuration. This configuration may include outbound registrations which
have been removed due to a reload operation occurring at the same time. The code for
printing the outbound registration did not take this into account but now it does.
AST-1506 #close
Review: https://reviewboard.asterisk.org/r/4338/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The res_pjsip module was previously unloadable. With this patch it can now
be unloaded.
This patch is based off the original patch on the issue (listed below) by Corey
Farrell with a few modifications. Namely, removed a few changes not required to
make the module unloadable and also fixed a bug that would cause asterisk to
crash on unloading.
This patch is the first step (should hopefully be followed by another/others at
some point) in allowing res_pjsip and the modules that depend on it to be
unloadable. At this time, res_pjsip and some of the modules that depend on
res_pjsip cannot be unloaded without causing problems of some sort.
The goal of this patch is to get res_pjsip and only res_pjsip to be able to
unload successfully and/or shutdown without incident (crashes, leaks, etc...).
Other dependent modules may still cause problems on unload.
Basically made sure, with the patch applied, that res_pjsip (with no other
dependent modules loaded) could be succesfully unloaded and Asterisk could
shutdown without any leaks or crashes that pertained directly to res_pjsip.
ASTERISK-24485 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4311/
patches:
pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code was missing the case for explicitly destroying an outbound publication
when Asterisk had never actually published anything. The result was that Asterisk
would hang for a while on a graceful shutdown.
With this change, the case is taken into account, and on a graceful shutdown, these
publications are destroyed without the need to actually send a PUBLISH request.
ASTERISK-24655 #close
Reported by Kevin Harwell
Review: https://reviewboard.asterisk.org/r/4325
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change makes the T.38 negotiation timeout configurable via
't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously
hard coded to be 5000 milliseconds.
This change also handles T.38 switch failures by aborting the fax since
in the case where this can happen, both sides have agreed to switch to
T.38 and Asterisk is unable to do so.
Review: https://reviewboard.asterisk.org/r/4320/
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Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you do a 'core (shutdown|restart) graceful' persistent subscriptions won't
survive. If you do a 'core (shutdown|restart) now' or asterisk terminates for
some reason, they do. Here's why...
When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to
subscribers for each subscription. This not only tells the subscribers that the
dialog/state machine is done, it also frees the last reference to the
subscription tree which causes the persistent subscription to get deleted from
astdb. When asterisk restarts, nothing's left. Just preventing the delete from
astdb doesn't work because we already told the subscriber to terminate the
dialog so we can't restart it even if it was still in astdb. Everything works
OK if asterisk terminates unexpectedly because we never send the 'terminated'
message so on restart, the subscription is still in astdb and the subscriber is
none the wiser.
This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for
persistent connections.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4318/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Every time a registration started, sip_outbound_registration_response_cb bumps
the ref count on client_state then pushes a handle_registration_response task.
handle_registration_response never unreffed it though. So every time a
registration goes out, the ref count goes up by one.
This patch adds the unreffs to handle_registration_response.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4303/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are 2 issues with reloading registrations...
1. The 'can_reuse_registration' test wasn't considering the intervals or
expiration in its determination of whether a registration changed or not so if
you changed any of the intervals or the expiration and reloaded, the object
would get reloaded but the actual timers wouldn't change.
can_reuse_registration now does a sorcery diff on the old and new objects
instead of discretely testing certain fields. Now if you change expiration for
instance, and reload, the timer is updated and re-registration will occur on the
new value.
2. If you mung up your password on an outbound registration you get a permanent
failure. If you fix the password (on the outbound_auth object) and reload,
nothing tells outbound_registration to try again because the registration itself
didn't change. This patch adds an observer on the "auth" object type and if any
auth changes, existing registration states are searched and those in a
REJECTED_PERMANENT state are retried.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4304/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the AMI Redirect action is used with a channel bridged inside
Stasis() and not running a pbx, the channel is hung up instead of
proceeding to the desired location in dialplan. This change allows
such channels to be Redirected properly by detecting the operation
used by Redirect (ASYNCGOTO) and using the code already established
for functionality of the ARI channel continue operation.
ASTERISK-24591 #close
Review: https://reviewboard.asterisk.org/r/4271/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this patch, the following two ARI commands
POST /channels
POST /channels/{id}/continue
Accept a new parameter, label, that can be used to continue to or originate
to a priority label in the dialplan.
Because this is adding a new parameter to ARI commands, the API version of
ARI has been bumped from 1.6.0 to 1.7.0.
This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks!
ASTERISK-24412 #close
Reported by Nir Simionovich
Review: https://reviewboard.asterisk.org/r/4285
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "MWI Subscription failed" message means the client is trying to subscribe
to a mailbox that doesn't exist. There's no need to clutter up logs with
warnings for a client misconfiguration so I changed it to a notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4306/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fix has two parts:
* Corrected an error message to properly state that external_replaces is an extension. The
error message also prints what dialplan context the external_replaces extension was being
looked for in.
* Corrected the printing of the Replaces: header in an INVITE request. We were duplicating
"Replaces: " in the header.
ASTERISK-24376 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/4296
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When res_pjsip loads and an endpoint auto-subscribes a mailbox for mwi,
if a contact hasn't registered yet, res_pjsip_mwi spits out a warning.
This is a perfectly normal situation though and doesn't require something
as serious as a warning. It's also self correcting. The device will start
getting mwi as soon as it registers.
This patch changes the warning to a notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4314/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The current behavior of 'pjsip send unregister' is to send the unregister
(REGISTER with 0 exp) but let the next scheduled register proceed normally.
I don't think that's a good idea. If you unregister, it should stay
unregistered until you decide to start registrations again. So this patch
just adds a cancel_registration call to the current unregister_task to
cancel the timer.
Of course, now you need a way to start registration again so I've added
a 'pjsip send register' command that unregisters and cancels any existing
registration (the same as send unregister), then sends an immediate
registration and starts the timer back up again.
Both changes also ripple to AMI. There's a new PJSIPRegister command.
There's no harm in calling either command repeatedly. They don't care
about the actual state.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4301/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some reason I was using a hash container instead of a list to gather the
contacts for 'pjsip list/show contacts' so even though I had a sort function,
the output wasn't sorted. This patch just changes the hash container to a
list container and the contacts now appear sorted in the CLI.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4305/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The PJSIP_AOR dialplan function allows inspection of configured AORs including
what contacts are currently bound to them.
The PJSIP_CONTACT dialplan function allows inspection of contacts in existence.
These can include both externally added (by way of registration) or permanent
ones.
ASTERISK-24341
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/4308/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This updates the documentation for the 'method' configuration option to
be more verbose about the behaviors of values 'unspecified' and
'default'. They do exactly the same thing which is to select the
default as defined by PJSIP which is currently TLSv1.
Review: https://reviewboard.asterisk.org/r/4264/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This backports the following from trunk, which were missed:
r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04 Nov 2014) | 2 lines
res_pjsip: Allow + at the beginning of a phone number when user_eq_phone is enabled.
r427259 | file | 2014-11-04 16:51:32 -0600 (Tue, 04 Nov 2014) | 2 lines
res_pjsip: Apply the 'user_eq_phone' setting to the To header as well.
It also adds the Alembic script for the option.
ASTERISK-24643
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note that this is backport from trunk of r425825.
This change adds a module which is configurable using the keep_alive_interval setting in the
global section that will send a CRLF keep alive to all active connection-oriented transports at
the provided interval. This is useful because it can help keep connections open through NATs.
This functionality also exists within PJSIP but can not be controlled at runtime and requires
recompiling it.
Review: https://reviewboard.asterisk.org/r/4084/
ASTERISK-24644 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you remove an endpoint/aor from pjsip.conf then do a core reload,
qualifies will continue even though the object are gone. This happens
because nothing clears out the qualify tasks.
This patch unschedules all existing qualify tasks before scheduling
new ones on reload.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4290/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reloading wasn't working correctly because on a reload, the sorcery apply
handler was never being called for unchanged users. So, instead of using
an apply handler, I'm now iterating over all users. Works much more reliably.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4288/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A native rtp bridge was being chosen (it shouldn't have been) when using two
pjsip channels with incompatible DTMF modes. This patch sets the rtp instance
property, AST_RTP_PROPERTY_DTMF, for the appropriate DTMF mode(s) for pjsip.
It was not being set before, meaning all DTMF modes for pjsip were being treated
as compatible, thus native bridging would be chosen as the bridge type when it
shouldn't have been.
ASTERISK-24459 #close
Reported by: Yaniv Simhi
Review: https://reviewboard.asterisk.org/r/4265/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, Asterisk would always respond to 401 responses to
registration attempts by trying to provide a registration with authentication
credentials. Even if subsequent attempts were rejected with 401 responses,
Asterisk would continue this behavior. If authentication credentials were
incorrect, this could continue forever.
With this patch, we keep track of whether we have attempted authentication
on an outbound registration attempt. If we already have, we don not try
again until the next attempt. This prevents the infinite loop scenario.
Review: https://reviewboard.asterisk.org/r/4273
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If certain pjsip modules aren't loaded, the wizard causes a SEGV
when it unloads. Added a check for the presense of the object
type wizard before trying to clean it up.
Tested-by: George Joseph
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
Those fixes included things like:
-out += sprintf(out, "%%%02X", (unsigned char) *ptr);
+out += sprintf(out, "%%%02X", (unsigned) *ptr);
That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.
This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)
Review: https://reviewboard.asterisk.org/r/4263/
ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a remote endpoint reinvites to T.38 immediately the state machine
will go into a peer reinvite state. If a T.38 capable application
(such as ReceiveFax) queries it will receive this state. Normally
the application will then indicate so that the channel driver will
queue up the T.38 offer previously received. Once it receives this
offer the application will act normally and negotiate.
The res_pjsip_t38 module incorrectly partially squashed this indication.
This would cause the application to think the request had failed when
in reality it had actually worked.
This change makes it so that no T.38 control frames (or indications)
are squashed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_pjsip_config_wizard
------------------
* This is a new module that adds streamlined configuration capability for
chan_pjsip. It's targetted at users who have lots of basic configuration
scenarios like 'phone' or 'agent' or 'trunk'. Additional information
can be found in the sample configuration file at
config/samples/pjsip_wizard.conf.sample.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4190/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Verbose message displayed when a file is played back via 'stream file'
was formatted differently than other playbacks:
* It didn't include the channel name
* It didn't include the channel language
It does, however, include the playback offset as well as any escape digits.
That information was kept; however, this patch updates the formatting to more
closely match the Verbose messages displayed when a file is played back by
'control stream file', Playback, ControlPlayback, or any other file playback
operation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a log message notifying the user that a stale configuration
is in place upon reload when a config object fails to load. This
situation can end up causing confusion when the object failed to load
but exists from a previous config load especially when the old config
is significantly different from the new config.
Review: https://reviewboard.asterisk.org/r/4250/
Reported by: Thomas Thompson
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Given the scenario where a PJSIP channel is in a native RTP bridge with direct
media and the channel is then hung up the code will currently re-INVITE the channel
back to Asterisk and send a BYE at the same time. Many SIP implementations dislike
this greatly.
This change makes it so that if a re-INVITE transaction is in progress the BYE
is queued to occur after the completion of the transaction (be it through normal
means or a timeout).
Review: https://reviewboard.asterisk.org/r/4248/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the past the SDP negotiation within res_pjsip_session was made more tolerant of
certain situations. The only case where SDP negotiation will fail is when a major
error occurs during negotiation. Receiving an already declined media stream is
not considered a major error.
When producing the local SDP the logic took this into account so on the initial INVITE
the declined media stream did not cause an SDP negotiation failure. Unfortunately
the logic for handling media streams with a handler did not mirror this logic and
considered an already declined media stream an error and thus failed the SDP
negotiation.
This change makes the logic between both situations match so only under major
errors will the SDP negotiation fail.
ASTERISK-24607 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4254/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Frames with a payload length of 0 were incorrectly handled in res_http_websocket.
Provided a frame with a payload had been received prior it was possible for a double
free to occur. The realloc operation would succeed (thus freeing the payload) but be
treated as an error. When the session was then torn down the payload would be
freed again causing a crash. The read function now takes this into account.
This change also fixes assumptions made by users of res_http_websocket. There is no
guarantee that a frame received from it will be NULL terminated.
ASTERISK-24472 #close
Reported by: Badalian Vyacheslav
Review: https://reviewboard.asterisk.org/r/4220/
Review: https://reviewboard.asterisk.org/r/4219/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When endpoints with direct_media enabled, behind a firewall (Asterisk on a
separate network) and were bridged sometimes Asterisk would send the ip
address of the firewall in the sdp to one of the phones in the reinvite
resulting in one way audio. When sending the reinvite Asterisk will retrieve
the media address from the associated rtp instance, but if frames were being
read this can be overwritten with another address (in this case the
firewall's). This patch ensures that Asterisk uses the original device
address when using direct media.
ASTERISK-24563
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4216/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using a non-default sorcery wizard (in this instance realtime) for outbound
publishes Asterisk will crash after a stack overflow occurs due to the code
infinitely recursing. The fix entails removing the outbound publish state
dependency from the outbound publish sorcery object and instead keeping an in
memory container that can be used to lookup the state when needed.
ASTERISK-24514 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4178/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an originator channel is specified when originating a channel the linked ID
of it will be applied to the newly originated outgoing channel. This allows
an association to be made between the two so it is known that the originator
has dialed the originated channel.
ASTERISK-24552 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4243/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The reviewboard description does a fine job of summarizing this, so here it is:
A reporter discovered that Asterisk would crash when attempting to retransmit
a reinvite that had previously received a 491 response. The crash occurred
because a pjsip_tx_data structure was being saved for reuse, but its reference
count was not being increased. The result was that the pjsip_tx_data was being
freed before we were actually done with it. When we attempted to re-use the
structure when re-sending the reinvite, Asterisk would crash.
The fix implemented here is not to try holding onto the pjsip_tx_data at all.
Instead, when we reschedule sending the reinvite, we create a brand new
pjsip_tx_data and send that instead. Because of this change, there is no need
for an ast_sip_session_delayed_request structure to have a pjsip_tx_data on
it any more. So any code referencing its use has been removed.
When this initial fix was introduced, I encountered a second crash when
processing a subsequent 200 OK on a rescheduled reinvite. The reason was
that when rescheduling the reinvite, we gave the wrong location for a
response callback. This has been fixed in this patch as well.
ASTERISK-24556 #close
Reported by Abhay Gupta
Review: https://reviewboard.asterisk.org/r/4233
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This corrects several bugs that currently exist in the stasis
application code.
* After a masquerade, the resulting channels have channel topics that
do not match their uniqueids
** Masquerades now swap channel topics appropriately
* StasisStart and StasisEnd messages are leaked to observer
applications due to being published on channel topics
** StasisStart and StasisEnd publishing is now properly restricted
to controlling apps via app topics
* Race conditions exist where StasisStart and StasisEnd messages due to
a masquerade may be received out of order due to being published on
different topics
** These messages are now published directly on the app topic so this
is now a non-issue
* StasisEnds are sometimes missing when sent due to masquerades and
bridge swaps into and out of Stasis()
** This was due to StasisEnd processing adjusting message-sent flags
after Stasis() had already exited and Stasis() had been re-entered
** This was corrected by adjusting these flags prior to sending the
message while the initial Stasis() application was still shutting
down
Review: https://reviewboard.asterisk.org/r/4213/
ASTERISK-24537 #close
Reported by: Matt DiMeo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When repeatedly starting/stopping a Monitor on a channel, the accumulated
in/out sample counts are never reset to 0. This can cause inadvertent jumps
in the recordings, as the code in the channel core will determine incorrectly
that a jump in the recorded file position should occur. Setting the sample
counts to 0 simply reflects the initial state a Monitor should be in when it
is started, as this is the initial count that would be on the channels at that
time.
ASTERISK-24573 #close
Reported by: Nuno Borges
patches:
24573.patch uploaded by Nuno Borges (License 6116)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are two methods within res_pjsip_refer for keeping track of the state of a transfer.
The first is a framehook which looks at frames passing by to determine the state. The second
subscribes to know when the channel joins a bridge. In the case when the channel joins the
bridge the framehook is *NOT* removed and this prevents the native RTP bridging technology
from getting used.
This change gets the channel and if it still exists remove the framehook.
Review: https://reviewboard.asterisk.org/r/4218/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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As a result of https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime
was tossing database fields that didn't have an exact match to a sorcery
registered field. This broke the ability to use regexes as field names which
manifested itself as a failure of res_pjsip_phoneprov_provider which uses
this capability. It also broke handling of fields that start with '@' in
realtime but I don't think anyone noticed.
This patch does the following...
* Modifies ast_sorcery_fields_register to pre-compile the name regex.
* Modifies ast_sorcery_is_object_field_registered to test the regex if it
exists instead of doing an exact strcmp.
* Modifies res_pjsip_phoneprov_provider with a few tweaks to get it to work
with realtime.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4185/
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The implementation of INVITE with Replaces in res_pjsip_refer did not expect them to
occur in-dialog. As a result it would incorrectly attempt to hang up a channel it
thought was under its control. In reality the channel would be under the control of
another thread. When the other thread accessed the channel it would be accessing freed
memory and could crash.
This change makes res_pjsip_refer not act on an in-dialog INVITE with Replaces.
ASTERISK-24528 #close
Reported by: Joshua Colp
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Optimistic SRTP is the ability to enable SRTP but not have it be
a fatal requirement. If SRTP can be used it will be, if not it won't be.
This gives you a better chance of using it without having your sessions
fail when it can't be.
Encrypt all the things!
Review: https://reviewboard.asterisk.org/r/3992/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is no guarantee that when we get a Refer-To that it will be NULL terminated.
As the URI parsing function requires it to be we now NULL terminate it.
Additionally parsing the Refer-To as a 'To' header is needless and it can
simply be done as a URI. This also fixes a problem where certain Refer-To headers
would not be parsed as a 'To' header causing the REFER to fail.
ASTERISK-24508 #close
Reported by: Beppo Mazzucato
Review: https://reviewboard.asterisk.org/r/4187/
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This change enforces the requirements in PJSIP for session timer configuration. The minimum
expiration period must be 90 seconds or higher and the normal expiration period can not
be lower than the minimum expiration period. If either of these were done the code would
assert at session setup time.
ASTERISK-24336 #close
Reported by: Leon Rowland
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From reviewboard:
"During blind transfer testing, it was noticed that tests were failing
occasionally because the ARI blind transfer event was not being sent.
After investigating, I detected a race condition in the blind transfer
code. When blind transferring a single channel, the actual transfer
operation (i.e. removing the transferee from the bridge and directing
them to the proper dialplan location) is queued onto the transferee
bridge channel. After queuing the transfer operation, the blind transfer
Stasis message is published. At the time of publication, snapshots of
the channels and bridge involved are created. The ARI subscriber to the
blind transfer Stasis message then attempts to determine if the bridge
or any of the involved channels are subscribed to by ARI applications.
If so, then the blind transfer message is sent to the applications. The
way that the ARI blind transfer message handler works is to first see
if the transferer channel is subscribed to. If not, then iterate over
all the channel IDs in the bridge snapshot and determine if any of
those are subscribed to. In the test we were running, the lone
transferee channel was subscribed to, so an ARI event should have been
sent to our application. Occasionally, though, the bridge snapshot did
not have any channels IDs on it at all. Why?
The problem is that since the blind transfer operation is handled by a
separate thread, it is possible that the transfer will have completed and
the channels removed from the bridge before we publish the blind transfer
Stasis message. Since the blind transfer has completed, the bridge on
which the transfer occurred no longer has any channels on it, so the
resulting bridge snapshot has no channels on it. Through investigation of
the code, I found that attended transfers can have this issue too for the
case where a transferee is transferred to an application."
The fix employed here is to decouple the creation of snapshots for the transfer
messages from the publication of the transfer messages. This way, snapshots
can be created to reflect what they are at the time of the transfer operation.
Review: https://reviewboard.asterisk.org/r/4135
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When using a non-default sorcery wizard (in this instance realtime) for
outbound registrations and after adding in an appropriate call to
ast_sorcery_apply_config() (since it is missing) Asterisk will crash after
a stack overflow occurs due to the code infinitely recursing. The fix entails
removing the outbound registration state dependency from the outbound
registration sorcery object and instead keeping an in memory container that
can be used to lookup the state when needed.
ASTERISK-24514
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4164/
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This change corrects message ordering in cases where a channel-related
message can be received after a Stasis/ARI application has received the
StasisEnd message. The StasisEnd message was being passed to
applications directly without waiting for the channel topic to empty.
As a result of this fix, other bugs were also identified and fixed:
* StasisStart messages were also being sent directly to apps and are
now routed through the stasis message bus properly
* Masquerade monitor datastores were being removed at the incorrect
time in some cases and were causing StasisEnd messages to not be sent
* General refactoring where necessary for the above
* Unsubscription on StasisEnd timing changes to prevent additional
messages from following the StasisEnd when they shouldn't
A channel sanitization function pointer was added to reduce processing
and AO2 lookups.
Review: https://reviewboard.asterisk.org/r/4163/
ASTERISK-24501 #close
Reported by: Matt Jordan
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When handling incoming messages we determine if it is associated with
a dialog. If so we use that to determine what serializer and endpoint
to use for the message. Previously this would pass the endpoint to the
endpoint lookup module to actually place the endpoint completely on the
message. For in-dialog responses, however, this did not occur as
dialog processing took over and the endpoint lookup did not occur.
This change just places the endpoint in the expected spot immediately
instead of relying on the endpoint lookup module. In-dialog responses
thus have the expected endpoint.
AST-1459 #close
Review: https://reviewboard.asterisk.org/r/4146/
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Testing has shown repeatedly that PJSIP's default behavior of switching
automatically to TCP for large messages can cause issues. The most common
issues are that devices that we are communicating with do not handle the
switch to TCP gracefully, thus causing situations such as broken calls or
broken subscriptions. Now, in order to have this behavior happen, you must
opt into it. The sample file has been updated to warn that enabling the
TCP switch behavior may cause issues for you, so use at your own risk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In
some circumstances (on some networks), this can cause some issues with
messages not getting sent to the correct destination - and can also cause
connections to get dropped due to quirks in pjproject deciding to
terminate TCP connections with no messages.
While fixing the routing/messaging issues is important, having a
configuration option in Asterisk that tells pjproject to not switch over
to TCP would be useful. That way, if some glitch is discovered on some
other network/site, we can at least disable the behavior until a fix is
put into place.
AFS-197 #close
Review: https://reviewboard.asterisk.org/r/4137/
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When the res_stasis module is unloaded, it will dispose of the apps_registry
container. This is a problem if an ARI operation is in flight that attempts
to use the registry, as the shutdown occurs in a separate thread. This patch
adds some sanity checks to the various routines that access the registry which
cause the operations to fail if the apps_registry does not exist.
Crash caught by the Asterisk Test Suite.
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A question arose as to whether a .pem file
could be provided in place of the .crt and
.key files in a PJSIP TLS configuration. I
tested this and discovered that although a
cert will be read from the pem file, a key
will not, and thus the priv_key_file entry
is still required. This update to the fine
documentation clarifies the option usage.
AST-1448 #close
Review: https://reviewboard.asterisk.org/r/4129/
Reported by: John Bigelow
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This updates the status of the outbound registration
to reflect when it has been unregistered. Since the
registration is unregistered but is not stopped, the
registration schedule remains active as before. The
patch also updates the documentation of both the AMI
and CLI commands.
ASTERISK-24411 #close
Review: https://reviewboard.asterisk.org/r/4119/
Reported by: John Bigelow
patches:
unregister-patch1.txt uploaded by John Bigelow (License 5091)
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Currently, it is possible for some subscriptions to get into a NULL state. When
this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a
device is subscribed for extension state then the associated subscription state
object can't be located. The code then attempts to dereference a NULL object.
Added a NULL check to avoid the problem.
Reported by: John Bigelow
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When removing the qualify_frequency from an AoR or a contact the statistics
shown when issuing "pjsip show aors" from the CLI are incorrect. This patch
deletes the contact's status object from sorcery, disassociating it from the
contact, if the qualify_freqency is removed from configuration.
ASTERISK-24462 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4116/
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George Joseph
Richard Mudgett
Matt Jordan
Mark Michelson
Scott Griepentrog
Corey Farrell
Kevin Harwell
Jonathan Rose
Scott Griepentrog
Joshua Colp
David M. Lee
Sean Bright
Walter Doekes
Ashley Sanders
Kinsey Moore