When SUBSCRIBE dialogs were established, we never associated
the endpoint that created the subscription with the dialog
we end up creating. In most cases, this ended up not causing
any problems.
The actual bug that was observed was that when a device that
was behind NAT established a subscription with Asterisk, Asterisk
would end up sending in-dialog NOTIFY requests to the device's
private IP addres instead of the public address of the NAT router.
When Asterisk receives the initial SUBSCRIBE from the device,
res_pjsip_nat rewrites the contact to the public address on which the
SUBSCRIBE was received. This allows for the dialog to have its target
address set to the proper public address. Asterisk then would send a 200
OK response to the SUBSCRIBE, then a NOTIFY with the initial
subscription state. The device would then send a 200 OK response to
Asterisk's NOTIFY.
Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat
did not rewrite the address in the Contact header. Then, when the PJSIP
dialog layer processed the 200 OK, PJSIP would perform a comparison
between the IP address in the Contact header and its saved target
address for the dialog. Since they differed, PJSIP would update the
target dialog address to be the address in the Contact header. From this
point, if Asterisk needed to send a NOTIFY to the device, the result was
that the NOTIFY would be sent to the private address that the device
placed in the Contact header.
The reason why res_pjsip_nat did not rewrite the address when it
received the 200 OK response was that it could not associate the
incoming response with a configured endpoint. This is because on a
response, the only way to associate the response to an endpoint is by
finding the dialog that the response is associated with and then finding
the endpoint that is associated with that dialog. We do not perform
endpoint lookups on responses. res_pjsip_pubsub skipped the step of
associating the endpoint with the dialog we created, so res_pjsip_nat
could not find the associated endpoint and therefore couldn't rewrite
the contact.
This commit message is like 50x longer than the actual fix.
ASTERISK 24981 #close
Reported by Mark Michelson
Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd
Contact status rtt is an int64_t and needs the PRId64 macro to
properly create the format specifier on 32-bit systems.
Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7
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
