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Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49
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According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0
was never returned historically and several users incorrectly coded usage
of the returned sched ID assuming that 0 was invalid.
ASTERISK-25476
Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20
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chan_sip.c:
* Initialize mwi subscription scheduler ids earlier because of ASTOBJ to
ao2 conversion.
* Initialize register scheduler ids earlier because of ASTOBJ to ao2
conversion.
chan_skinny.c:
* Fix more scheduler usage for the valid 0 id value.
ASTERISK-25476
Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95
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channels/chan_iax2.c:
* Initialize struct chan_iax2_pvt scheduler ids earlier because of
iax2_destroy_helper().
channels/chan_sip.c:
channels/sip/config_parser.c:
* Fix initialization of scheduler id struct members. Some off nominal
paths had 0 as a scheduler id to be destroyed when it was never started.
chan_skinny.c:
* Fix some scheduler id comparisons that excluded the valid 0 id.
channel.c:
* Fix channel initialization of the video stream scheduler id.
pbx_dundi.c:
* Fix channel initialization of the packet retransmission scheduler id.
ASTERISK-25476
Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8
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Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7
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The fastagi record-file testsuite test sometimes fails reporting an empty
recorded file. This was happening because Asterisk was sending the agi result
notification prior to actually closing the file and the data, being buffered,
had not been written to the file yet when the test attempts to check the file
size.
This patch makes it so the record file stream is closed prior to sending the
agi result notification.
ASTERISK-25593 #close
Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde
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Several issues are addressed here:
- main() is large, and half of it is only used if we're not rasterisk;
fixed by spliting up the daemon part into a separate function.
- Call ast_term_init from rasterisk as well.
- Remove duplicate code reading/writing asterisk history file.
- Attempt to tackle background color issues and color changes that
occur. Tested by starting asterisk -c until the colors stopped
changing at odd locations.
ASTERISK-25585 #close
Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f
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Fixes some minor typos in the CHANGES file, plus an embarrasing typo in
the StatsD API.
Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7
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The usage info for 'pjsip send notify' previously referenced the
chan_sip configuration sip_notify.conf. Fix this to reference
the correct configuration pjsip_notify.conf.
ASTERISK-25590 #close
Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea
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negotiation." into 13
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If the sorcery object type is not found a NULL is returned.
Unfortunately, sorcery_realtime_filter_objectset() will crash after
complaining about not finding the object type and saying to expect errors.
* Use ao2_cleanup() instead of ao2_ref() to prevent the crash.
ASTERISK-25165
Reported by Corey Farrell
Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97
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statistics" into 13
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into 13
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StatsD" into 13
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When a channel is in a direct media bridge, a re-INVITE may arrive that forces
Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge
must change its technology to a simple bridge, and re-INVITE the media back
to Asterisk.
Generally, this logic mostly already exists in Asterisk. However, prior to this
patch, there were a few bugs:
(1) The T.38 framehook currently prevents a channel capable of T.38 faxes from
ever entering into a direct media bridge. This applies even when the only
media being passed over the channel is audio. This patch fixes this bug
by having the framehook specify that it defers caring about any frame type.
This allows the channels to enter into a direct media bridge, which will
be broken when a re-INVITE is received.
(2) When a re-INVITE is received, nothing instructed the bridging layer to
re-inspect the allowed bridging technology. This now occurs when either
a re-INVITE is received from a peer, or when a response is received from
the far end (that is, when the T.38 state changes to either
T38_PEER_REINVITE or T38_LOCAL_REINVITE).
(3) chan_pjsip needs to do a small amount of work to prevent a direct media
bridge from being chosen when a T.38 session is in progress. When a T.38
session supplement has a t38 datastore - which is added when we detect
we should start thinking about T.38 on a channel - we now refuse a native
RTP bridge.
(4) When a BYE request is received, we don't terminate the T.38 session. If
the other side of a T.38 fax survives the hangup (due to the 'g' flag
in Dial, for example), we don't currently re-INVITE the media on the
other channel back to audio. This patch now has res_pjsip_t38 intercept
BYE requests and inform the far side that the T.38 session is terminated.
This naturally causes the correct re-INVITEs to be sent.
ASTERISK-25582
Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb
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context/exten/app" into 13
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timeouts." into 13
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This patch adds some debug statements to res_pjsip_t38. These statements help
to determine which SDP negotiation callbacks are being executed, and, when
a particular callback exits, why a callback may not have applied its logic
to the local or remote SDP.
Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77
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Because the context, extension, and application are stored in stringfields,
checking for them being NULL doesn't work so well. This patch uses the
appropriate string library call, ast_strlen_zero, to see if there is a value
in the context/exten/app values.
Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23
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This patch adds a module that emits StatsD statistics about Asterisk
endpoints. This includes:
* A GUAGE statistic for endpoint states, tracking how many endpoints are in
a particular state.
* A GUAGE statistic for each endpoint, counting the number of channels
currently associated with an endpoint.
ASTERISK-25572
Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305
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This patch adds the ability to send StatsD statistics related to the
state of PJSIP contacts. This includes:
* A GUAGE statistic measuring the count of contacts in a particular state.
This measures how many contacts are reachable, unreachable, etc.
* The RTT time for each contact, if those contacts are qualified. This
provides StatsD engines useful time-based data about each contact.
ASTERISK-25571
Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c
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This patch adds outbound registration statistics for StatsD. This includes
the following:
* A GUAGE metric for the overall count of outbound registrations.
* A GUAGE metric for each state an outbound registration can be in. As the
outbound registrations change state, the overall count of how many
outbound registrations are in the particular state is changed.
These statistics are particularly useful for systems with a large number of
SIP trunks, and where measuring the change in state of the trunks is useful
for monitoring.
ASTERISK-25571
Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37
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When Asterisk is configured to use a dynamic sorcery backend (such as
res_sorcery_astdb) with 'registration' objects, it will fail to create the
internal state objects associated with the registration objects on module
load. This is due to nothing actually querying for the specific objects
and calling their sorcery apply handler during module load.
This patch fixes that by calling get_registrations in the sorcery observer's
object_type_loaded handler. Doing this causes the sorcery backends to be
asked for the current state of all registration objects, which causes the
apply handler to be called and the internal run-time state to be created.
ASTERISK-25575 #close
Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23
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Previously, a trancoding module did not have access to the joint but cached
format. Therefore, the module did not have access to the attributes negotiated
via SDP (line fmtp). Now, a translation module receives the joint format.
ASTERISK-25545 #close
Change-Id: Id6878a989b50573298dab115d3371ea369e1a718
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When no parameter is present, Asterisk does not generate the line fmtp, as
expected. However, because a buffer was reset, even rtpmap and fmtp of previous
media codecs got removed. Now, Asterisk does not reset other codecs in case of
no parameter for H.264.
ASTERISK-25573 #close
Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286
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Often, the metric names of statistics we are generating for StatsD have some
dynamic component to them. This can be the name of a particular resource, or
some internal status label in Asterisk. With the current set of functions,
callers of the statsd API must first build the metric name themselves, then
pass this to the API functions. This results in a large amount of boilerplate
code and usage of either fixed length static buffers or dynamic memory
allocation, neither of which is desireable.
This patch adds two new functions to the StatsD API that support a printf
style format specifier for constructing the metric name. A dynamic string,
allocated in threadstorage, is used to build the metric name. This eases
the burden on users of the StatsD API.
Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea
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Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d
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Receiving a 423 Interval Too Brief response after authentication for an
outbound registration attempt results in assuming that the registrar has
rejected the registration permanently. If there are no configured retries
for fatal responses then the outbound registration is stopped for that
endpoint.
For registrations, PJSIP/PJPROJECT intercepts the handling of 423
responses and does not include any authentication in the updated
registration request. When the updated request is challenged then the
Asterisk code assumes that we were challenged again because the peer
rejected the authentication we sent earlier.
* Made registration challenges keep track of the CSeq number to determine
if the received challenge response was for the request we thought we sent.
If the response's CSeq number differs from the CSeq number we last sent
with authentication then authenticate again because it is a challenge to a
different request.
Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09
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Added a new api to res_statsd.c to allow it to receive a
character pointer for the value argument. This allows for a
'+' and a '-' to easily be sent with the value.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611
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a timer" into 13
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When a request is sent using pjsip_endpt_send_request and fails, a condition
exists where the request wrapper, which is an AO2 object, may be de-ref'd
more times than it should. This occurs when the request's callback is called,
and, in the callback, the timer on the PJSIP heap is cancelled. When that
occurs, the request wrapper's lifetime is decremented. When
pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of
the request wrapper again, even though we've already cancelled the reference
associated with the timer.
This patch checks the return result of pj_timer_heap_cancel_if_active before
removing the reference associated with the timer. We now only decrement it
in this case if a timer is cancelled as a result of the function call.
Change-Id: I21332343a1a019c1117076f9bf2df27be2850102
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This option adds the ability to specify a timeout, in seconds, for a
participant in a ConfBridge. When the user's timeout has been reached,
the user is ejected from the conference with the CONFBRIDGE_RESULT
channel variable set to "TIMEOUT".
The rationale for this change is that there have been times where we
have seen channels get "stuck" in ConfBridge because a network issue
results in a SIP BYE not being received by Asterisk. While these
channels can be hung up manually via CLI/AMI/ARI, adding some sort of
automatic cleanup of the channels is a nice feature to have.
ASTERISK-25549 #close
Reported by Mark Michelson
Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98
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commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525)
refer ASTERISK-24958
above commit removed ast_channel_lock(qe->chan);
but failed to remove corresponding ast_channel_unlock(qe->chan);
ASTERISK-25561 #close
Reported Alec Davis
Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a
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The hashtab API is pretty NULL tolerant which has resulted
in remaining callers not doing much checks themselves.
Unfortunately the function to destroy an iterator does not
do a NULL check and will result in a crash if passed NULL.
This change fixes that.
ASTERISK-25552 #close
Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619
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If an authenticated incoming caller does not respond to our 200 OK INVITE
response with an ACK then PJSIP will hangup the call. Unfortunately,
there is a chance that the session's channel will go away between one use
of the channel pointer and another when building the BYE request because
the BYE is being built by the monitor thread and not the call's serializer
thread.
* Added a check to ensure that the thread trying to add the Reason header
is the call's serializer thread. This ensures that the channel will not
go away on us.
Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89
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In practical tests, we have seen certain taskprocessors, specifically
Stasis subscription taskprocessors, cross the recently-added high-water
mark and emit a warning. This high-water mark warning is only intended
to be emitted when things have tanked on the system and things are
heading south quickly. In the practical tests, the Stasis taskprocessors
sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in
any danger at all.
As such, this ups the high-water mark to 500 tasks instead. It also
redefines the SIP threadpool request denial number to be a multiple of
the taskprocessor high-water mark.
Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce
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In Asterisk 13, cached formats are created before their corresponding format-
attribute module is registered. Cached formats are involved when a local
extension is called. Therefore, ast_format_generate_sdp_fmtp did not work
on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264,
and format-attribute modules provided externally.
ASTERISK-25160 #close
Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354
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