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An http request can be sent to delete a log channel
in Asterisk.
The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/logging/mylog'" can be run in the terminal
to access the newly implemented functionally for ARI.
* Able to delete log channels using ARI
ASTERISK-25252
Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6
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* changes:
rtp_engine.h: No sense allowing payload types larger than RFC allows.
rtp_engine.c: Minor tweaks.
rtp_engine.h: Misc comment fixes.
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An http request can be sent to rotate a specified log channel.
If the channel does not exist, an error response will be
returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/logging/logChannelName/rotate'" can be run in the
terminal to access this new functionality.
* Added the ability to rotate log files through ARI
ASTERISK-25252
Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01
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Prior to ASTERISK-24988, the WebSocket handshake was resolved before Stasis
applications were registered. This was done such that the WebSocket would be
ready when an application is registered. However, by creating the WebSocket
first, the client had the ability to make requests for the Stasis application
it thought had been created with the initial handshake request. The inevitable
conclusion of this scenario was the cart being put before the horse.
ASTERISK-24988 resolved half of the problem by ensuring that the applications
were created and registered with Stasis prior to completing the handshake
with the client. While this meant that Stasis was ready when the client
received the green-light from Asterisk, it also meant that the WebSocket was
not yet ready for Stasis to dispatch messages.
This patch introduces a message queuing mechanism for delaying messages from
Stasis applications while the WebSocket is being constructed. When the ARI
event processor receives the message from the WebSocket that it is being
created, the event processor instantiates an event session which contains a
message queue. It then tries to create and register the requested applications
with Stasis. Messages that are dispatched from Stasis between this point and
the point at which the event processor is notified the WebSocket is ready, are
stashed in the queue. Once the WebSocket has been built, the queue's messages
are dispatched in the order in which they were originally received and the
queue is concurrently cleared.
ASTERISK-25181 #close
Reported By: Matt Jordan
Change-Id: Iafef7b85a2e0bf78c114db4c87ffc3d16d671a17
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A testsuite test recently failed due to a crash that occurred in the DNS
core. The problem was that the test could not resolve an address, did
not set a result on the DNS query, and then indicated the query was
completed. The DNS core does not handle the case of a query with no
result gracefully, and so there is a crash.
This changeset makes the DNS system resolver set a result with a
zero-length answer in the case that a DNS resolution failure occurs
early. The DNS core now also will accept such a response without
treating it as invalid input. A unit test was updated to no longer treat
setting a zero-length response as off-nominal.
Change-Id: Ie56641e22debdaa61459e1c9a042e23b78affbf6
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* Tweaked add_static_payload() to not use magic numbers.
Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b
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Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43
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This will add ECDH support to Asterisk. It will
detect auto ECDH support in OpenSSL
(1.0.2b and above) during ./configure. If this is
available, it will use it,
otherwise it will fall back to prime256v1 (this
behavior is consistent with
other projects such as Apache and nginx).
This fixes WebRTC being broken in Firefox 38+ due
to Firefox now only supporting
ciphers with perfect forward secrecy.
ASTERISK-25265 #close
Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b
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This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
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This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
chan_sip option, this specifies an interval, in seconds, at which we
will send RTP comfort noise frames. This can be useful for keeping RTP
sessions alive as well as keeping NAT associations alive during lulls.
ASTERISK-25242 #close
Reported by Mark Michelson
Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d
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cache"
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Fixes for issues with the ASTERISK-24934 patch.
* Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is
an empty string. If it were an empty string the functions returned NULL
as if there were a memory allocation failure. This failure caused the AMI
VarSet event to not get posted if the new value was an empty string.
* Fixed dest buffer overwrite potential in ast_escape() and
ast_escape_c(). If the dest buffer size is smaller than the space needed
by the escaped s parameter string then the dest buffer would be written
beyond the end by the nul string terminator. The num parameter was really
the dest buffer size parameter so I renamed it to size.
* Made nul terminate the dest buffer if the source string parameter s was
an empty string in ast_escape() and ast_escape_c().
* Updated ast_escape() and ast_escape_c() doxygen function description
comments to reflect reality.
* Added some more unit test cases to /main/strings/escape to cover the
empty source string issues.
ASTERISK-25255 #close
Reported by: Richard Mudgett
Change-Id: Id77fc704600ebcce81615c1200296f74de254104
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An http request can be sent to reload an Asterisk module. If the
module can not be reloaded or is not already loaded, an error
response will be returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, based
on configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be reloaded through http requests
ASTERISK-25173
Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1
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An http request can be sent to retrieve information on a single
module, including the resource name, description, use count, status,
and support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending on
configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on a single module can now be retrieved
ASTERISK-25173
Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463
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During an attended transfer a thread is started that handles imparting the
bridge channel. From the start of the thread to when the bridge channel is
ready exists a gap that can potentially cause problems (for instance, the
channel being swapped is hung up before the replacement channel enters the
bridge thus stopping the transfer). This patch adds a condition that waits
for the impart thread to get to a point of acceptable readiness before
allowing the initiating thread to continue.
ASTERISK-24782
Reported by: John Bigelow
Change-Id: I08fe33a2560da924e676df55b181e46fca604577
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This patch adds a new API to the Asterisk core that acts as a media
cache. The core API itself is mostly a thin wrapper around some bucket
API provided implementation that itself acts as the mechanism of
retrieval for media. The media cache API in the core provides the
following:
* A very thin in-memory cache of the active bucket_file items. Unlike a
more traditional cache, it provides no expiration mechanisms. Most
queries that hit the in-memory cache will also call into the bucket
implementations as well. The bucket implementations are responsible
for determining whether or not the active record is active and valid.
This makes sense for the most likely implementation of a media cache
backend, i.e., HTTP. The HTTP layer itself is the actual arbiter of
whether or not a record is truly active; as such, the in-memory cache
in the core has to defer to it.
* The ability to create new items in the media cache from local
resources. This allows for re-creation of items in the cache on
restart.
* Synchronization of items in the media cache to the AstDB. This
also includes various pieces of important metadata.
The API provides sufficient access that higher level APIs, such as the
file or app APIs, do not have to worry about the semantics of the bucket
APIs when needing to playback a resource.
In addition, this patch provides unit tests for the media cache API. The
unit tests use a fake bucket backend to verify correctness.
Change-Id: I11227abbf14d8929eeb140ddd101dd5c3820391e
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This patch adds a new function to the bucket API for ast_bucket_file
objects, ast_bucket_file_metadata_callback. It will call ao2_callback on
the ast_bucket_file's ao2_container of metadata, calling the provided
ao2_callback_fn callback on each piece of metadata associated with the
file.
This is particularly useful when a bucket backend has added metadata,
and a higher level API wants to be aware of/access said metadata,
without knowing for sure what the key is.
Change-Id: I96f6757717f47b650df91a437f7df16406227466
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An http request can be sent to retrieve a list of all existing modules,
including the resource name, description, use count, status, and
support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/
asterisk/modules" (or something similar, depending on configuration)
can be run in the terminal to access this new functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on modules can now be retrieved
Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0
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Prior to this patch, the DNS core present in master had no default system-level
resolver implementation. Therefore, it was not possible for the DNS core to
perform resolutions unless the libunbound library was installed and the
res_resolver_unbound module was loaded.
This patch introduces a system-level DNS resolver implementation that will
register itself with the lowest consideration priority available (to ensure
that it is to be used only as a last resort). The resolver relies on low-level
DNS search functions to perform a rudimentary DNS search based on a provided
query and then supplies the search results to the DNS core.
ASTERISK-25146 #close
Reported By: Joshua Colp
Change-Id: I3b36ea17b889a98df4f8d80d50bb7ee175afa077
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When res_pjsip body generator modules were generating XML or XPIDF
response bodies, there was a chance that the generated body would be the
exact size of the supplied buffer. Adding the nul string terminator would
then write beyond the end of the buffer and potentially corrupt memory.
* Fix MALLOC_DEBUG high fence violations caused by adding a nul string
terminator on the end of a buffer for XML or XPIDF response bodies.
* Made calls to pj_xml_print() safer if the XML prolog is requested. Due
to a bug in pjproject, the return value could be -1 _or_
AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough.
* Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the
return value of pj_xml_print() when the supplied buffer is not large
enough.
ASTERISK-25168
Reported by: Carl Fortin
Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de
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Previously Asterisk did not properly failover to the next resolved DNS
address when a endpoint could not be reached. With this patch, and while
using res_pjsip, SIP requests (both in/out of dialog) now attempt to use
the next address in the list of resolved addresses until a proper response
is received or no more addresses are left.
ASTERISK-25076 #close
Reported by: Joshua Colp
Change-Id: Ief14f4ebd82474881f72f4538f4577f30af2a764
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This patch enhances the bucket API in two ways.
First, since ast_bucket and ast_bucket_file instances are immutable, a 'clone'
operation has been added that provides a 'clone' of an existing
ast_bucket/ast_bucket_file object. Note that this makes use of the
ast_sorcery_copy operation, along with the copy callback handler on the
"bucket" and "file" object types for the bucket sorcery instance.
Second, there is a need for the bucket API to ask a wizard if an object
is stale. This is particularly useful with the upcoming media cache
enhancements, where we want to ask the backing data storage if the
object we are currently operating on has known updates. This patch adds
API calls for ast_bucket and ast_bucket_file objects, which callback
into their respective sorcery wizards via the sorcery API.
Unit tests have also been added to cover the respective
ast_bucket/ast_bucket_file clone and staleness operations.
Change-Id: Ib0240ba915ece313f1678a085a716021d75d6b4a
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This patch enhances the sorcery API to allow for sorcery wizards to
determine if an object is stale. This includes the following:
* Sorcery objects now have a timestamp that is set on creation. Since
sorcery objects are immutable, this can be used by sorcery wizards to
determine if an object is stale.
* A new API call has been added, ast_sorcery_is_stale. This API call
queries the wizards associated with the object, calling a new callback
function 'is_stale'. Note that if a wizard does not support the new
callback, objects are always assumed to not be stale.
* Unit tests have been added that cover the new API call.
Change-Id: Ica93c6a4e8a06c0376ea43e00cf702920b806064
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A module trying to unload needs to wait for all serializers it creates and
uses to complete processing before unloading.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059
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Find and unlink the specified sorcery object type to complement
ast_sorcery_object_register(). Without this function you cannot
completely unload individual modules that use sorcery for configuration.
ASTERISK-24907
Reported by: Kevin Harwell
Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88
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Added checks when a unit test is registered to see that the summary and
description strings do not end with a new-line '\n' for consistency.
The check generates a warning message and will cause the
/main/test/registrations unit test to fail.
* Updated struct ast_test_info member doxygen comments.
Change-Id: I295909b6bc013ed9b6882e85c05287082497534d
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The res_pjsip_mwi previously required a reload to set up the proper
subscriptions to allow unsolicited MWI to work. This change
makes it so the act of registering will also cause this to occur.
This is particularly useful if realtime is involved as no reload
needs to occur within Asterisk to cause the MWI information
to get sent.
ASTERISK-25180 #close
Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252
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Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
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All send/receive processing for a SIP transaction needs to be done under
the same threadpool serializer to prevent reentrancy problems inside
pjproject when using an external DNS resolver to process messages for the
transaction.
* Add threadpool API call to get the current serializer associated with
the worker thread.
* Pick a serializer from a pool of default serializers if the caller of
res_pjsip.c:ast_sip_push_task() does not provide one.
This is a simple way to ensure that all outgoing SIP request messages are
processed under a serializer. Otherwise, any place where a pushed task is
done that would result in an outgoing out-of-dialog request would need to
be modified to supply a serializer. Serializers from the default
serializer pool are picked in a round robin sequence for simplicity.
A side effect is that the default serializer pool will limit the growth of
the thread pool from random tasks. This is not necessarily a bad thing.
* Made pjsip_resolver.c use the requesting thread's serializer to execute
the async callback.
* Made pjsip_distributor.c save the thread's serializer name on the
outgoing request tdata struct so the response can be processed under the
same serializer.
ASTERISK-25115 #close
Reported by: John Bigelow
Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a
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Change-Id: Icafea3fb4ea64ac027561b23cbfe2b17997dc549
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Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976
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Although ast_context_find, ast_context_find_or_create and
ast_context_destroy perform locking of the contexts table,
any context pointer can become invalid at any time that the
contexts table is unlocked. This change adds locking around
all complete operations involving these functions.
Places where ast_context_find was followed by ast_context_destroy
have been replaced with calls ast_context_destroy_by_name.
ASTERISK-25094 #close
Reported by: Corey Farrell
Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa
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So this issue is a bit complicated. Since it is possible to pass values to AMI
that contain a '\r\n' (or other similar sequences) these values need to be
escaped. One way to solve this is to escape the values and then pass the escaped
values to the AMI variable parameter string building function. However, this
puts the onus on the pre-build function to escape all string values. This
potentially requires a fair amount of changes along with a lot of string
allocations/freeing for all values.
Surely there is a way to push this complexity down a level into the string
building function itself? This of course is possible, but ends up requiring a
way to distinguish between strings that need to be escaped and those that don't.
The best way to handle this is by introducing a new format specifier in the
format string. For instance a %s (no escape) and %S (escape). However, that is
a bit weird and unexpected.
So faced with those possibilities this patch implements a limited version of the
first option. Instead of attempting to escape all string values this patch only
escapes those values that make sense. This approach limits the number of changes
and doesn't suffer from the odd format specifier problem.
ASTERISK-24934 #close
Reported by: warren smith
Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0
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The length of frames retured by sample functions was twice as large as
real, what caused global buffer overflow caught by AddressSanitizer.
ASTERISK-24717 #close
Reported by: Badalian Vyacheslav
Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6
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This reverts commit 6fca75bb628dfff2ab112e80b0228cf3ac0b8a05.
Change-Id: Ifee026cc63e22c5ac5717c37867a9f036373ae5a
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When an endpoint was created, it's messages were being forwarded to
both the tech endpoint topic and the all endpoints topic. Since
the tech topic was also forwarded to all, this was resulting in
duplicate messages whenever an endpoint published. This patch
causes the endpoint to only forward to the tech topic and lets
the tech topic forward to all.
To accomplish this, the existing stasis_cp_single_create function
(which both creates and forwards) was cloned and split into 2
functions, one that creates the topic and one that sets up the
forwarding. This allows endpoint_internal_create to create
the topic from the endpoint_all cache without forwarding it there,
then allows it to do the forward to the tech's topic.
ASTERISK-25137 #close
Reported-by: Vitezslav Novy
ASTERISK-25116 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c
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Add a new ContactStatus AMI event.
Publish the following status/state changes:
Created
Removed
Reachable
Unreachable
Unknown
Contact URI, new status/state, aor and endpoint names, and the
last qualify rtt result are included in the event.
ASTERISK-25114 #close
Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
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Many uses of stasis_unsubscribe in modules can be reached through unload.
These have been switched to stasis_unsubscribe_and_join.
Some subscription callbacks do nothing, for these I've created a noop
callback function in stasis.c. This is used by some modules that monitor
MWI topics in order to enable cache, since the callback does not become
invalid after dlclose it is safe to use stasis_unsubscribe on these, even
during module unload.
ASTERISK-25121 #close
Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c
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