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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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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"
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timeouts."
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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 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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type load"
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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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To be able to barge into a call by dialling a prefix+extension that maps
to the extensions device.
Senario is that DECT headset users may be away from their desks and need
to transfer the call, the goal is that from any phone they dial a prefix
then their extension and are added to the bridge that they are in, from
there they can drop the headset call, as it's also on the handset,
and transfer the caller.
The dialplan would look like, where prefix=73, extension = 8512;
exten => _738512,1,BridgeAdd(SIP/cisco0001)
ASTERISK-25551 #close
Reported By: Alec Davis
Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540
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Implemented support for the StatsD sample rate parameter,
which is a parameter for determining when to send computed
statistics to a client.
Valid sample rate values are:
Less than or equal to 0.0 will never be sent.
Between 0.0 and 1.0 will randomly be sent.
Greater than or equal to 1.0 will always be sent.
ASTERISK-25419
Reported By: Ashley Sanders
Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52
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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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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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a timer"
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When dns_parse_answer_ex was iterating over the answers it
wasn't incrementing the answer pointer correctly after the first
answer. The result was that no answers after the first
were being returned. For results where multiple records should
have been sorted by priority, weight, etc., there was nothing
to sort so the only the first record was returned even if it
wouldn't have been the correct record based on the sort.
ASTERISK-25565 #close
Reported-by: Daniel Tryba
Tested-by George Joseph
Change-Id: I8622604fefdcd3c11e2c5609a6382e53b1467b0b
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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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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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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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When the SIP threadpool is backed up with tasks, we send 503 responses
to ensure that we don't try to overload ourselves. The problem is that
we were not insuring that we were not trying to send a 503 to an
incoming SIP response.
This change makes it so that we only send the 503 on incoming requests.
Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404
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We have observed situations where the SIP threadpool may become
deadlocked. However, because incoming traffic is still arriving, the SIP
threadpool's queue can continue to grow, eventually running the system
out of memory.
This change makes it so that incoming traffic gets rejected with a 503
response if the queue is backed up too much.
Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816
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When appending all formats of a type all the codecs are iterated
and added. This operation was incorrectly adding the ast_format_none
format which is special in that it is supposed to be used when no
format is present. It shouldn't be appended.
ASTERISK-25535
Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c
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When ASTERISK-25449 was closed, a number of scheduler issues mentioned in
the comments were missed. These have since beed raised in ASTERISK-25476
and elsewhere.
This patch attempts to collect all of the scheduler issues discovered so
far and address them sensibly.
ASTERISK-25476 #close
Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b
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This change adds handling of dead worker threads when moving them
to be active. When this happens the worker thread is removed from
both the active and idle threads container. If no threads are able
to be moved to active then the pool grows as configured.
A unit test has also been added which thrashes the idle timeout
and thread activation to exploit any race conditions between the
two.
ASTERISK-25546 #close
Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143
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Previously, format-attribute modules relied on an existing fmtp line in SDP
negotiation. However, fmtp is optional for several formats like the Opus Codec.
Now, the format-attribute module is called with an empty fmtp, which allows the
module to initialise itself to RFC defaults. Furthermore now, Asterisk is able
to differentiate between internally and externally created formats.
ASTERISK-25537 #close
Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52
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increased."
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ABI compatibility stubs existed for ast_app_separate_args and ast_verbose,
this is not needed in master.
Change-Id: I07b4d2c16079da3c2c6efa55df4a74368e0bd453
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