* Eliminated RAII_VAR in get_outbound_endpoint().
* Simplify update_to() coding. However, this function can only be a NoOp
because the To string can only be a URI and not a name-address formatted
string.
* Simplify update_from() coding. Also fixed a code path modifying the
from string when the caller could still want to use the original string.
* Fixed msg_data_create() incompletely removing the "pjsip:" to then add
back the "sip:" string if needed. The code didn't handle the "pjsip:sip:"
case because it left the colon after pjsip in the string.
Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db
Currently when you add global headers from the dialplan both
the header in the dialplan and the globally configured header
are added to the resulting SIP INVITE. This change makes it
so the headers in the dialplan take precedence and are the
only ones added.
Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad
The code was referencing the config section as 'globals'
instead of 'general'. This change swaps it over to 'general'.
Change-Id: I9dfe7788f41c4a6754c77e103880dc1a747de7fe
Prior to this patch, a stop issued by a delete of a Playback resource
(indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop
the current media URI playing. Subsequent URIs specified by a playback
operation would then proceed on, even though we had just indicated to
the User that the Playback was finished *and* after they had just
'deleted' the resource. Whoops.
This patch corrects it by bailing out of the sequence of URIs to play if
one of them is terminated with an AST_CONTROL_STREAM_STOP indication.
ASTERISK-26341 #close
Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42
The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
http://downloads.digium.com/pub/telephony/ are now listed in the
"External" sections of the "Resource Modules" and "Codec Translators"
pages in menuselect. Any that are selected will automatically be
downloaded and installed when "make install" is run. Their LICENSE and
README (if avaialble) files will be installed to
ASTVARLIBDIR/documentation/thirdparty/<product_name>.
Example use with codecs:
The codecs/codecs.xml file is a menuselect style xml file that lists
the codecs to be included. Their support levels are 'external', which
triggers the download and install, and defaultenabled is no. Also
because codec_g729a is actually in a directory named codec_g729 on the
download server, the newly added 'member_data' element is used to
override the default of the directory name being the package name. You
can use the 'directory_name' attribute to keep default base URL
(http://downloads.digium.com/pub/telephony/) but use the new directory,
or you use the 'remote_url' attribute to specify a full URL to the
download directory. In this case, you must still follow the same
subdirectory naming conventions as that used for the packages located
at 'http://downloads.digium.com/pub/telephony'.
A new configure option '--with-externals-cache' was added and like
'--with-sounds-cache' it allows the installer to cache tarballs so
they're not downloaded every time.
To assist with the download and install process, each external package
now has a manifest.xml file that, among other things, contains a package
version and checksums for each file in the tarball. The manifest is
saved to both the cache directory and ASTMODDIR and together with the
manifest.xml on the downloads site, tells the install scripts whether
a download and/or update is needed.
bash and xmlstarlet are required for downloader operation. If they're
not installed, the external items in menuselect will be unavailable.
Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a
On heavy loaded system the TCP/TLS incoming calls could be
disconnected by pjproject while these calls are being
processed by asterisk which could use the session's memory pools.
If the session in the disconnected state then the session memory
pools were already freed, so we get segfault.
This patch adds a lifetime control on an INVITE session to pjproject.
The lifetime of the session is manipulated by calling
pjsip_inv_add_ref/pjsip_inv_dec_ref.
This patch uses these functions to inform pjproject that the
session is in use.
This patch adds check if the session state is not disconnected
and also checks if the memory pool is not NULL.
This patch also places tasks 'session_end' and 'session_end_completion'
into session's serializer to avoid race condition.
ASTERISK-26291 #close
Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7
Create an alternative to ast_sorcery_generic_alloc which uses astobj2
shared locking. Use this new method for the 'struct ast_sip_aor' allocator.
Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f
In ARI, the channels API allows to hangup a channel with a hangup reason.
This commit adds a new reason "answered_elsewhere".
When using a SIP channel, this will eventually allow Asterisk to add a proper
"Reason" header to a CANCEL message.
ASTERISK-26321
Change-Id: Ia97675bd4acd6a7f58eb467953dfb94559f6583d
If the PJSIP endpoint's AOR with the permanent contact
was deleted from the realtime storage the res_pjsip module
continues trying to qualify this contact.
The error 'Unable to find an endpoint to qualify contact'
appeares every 'qualify_frequency' seconds.
This patch deletes this contact in this case.
The PJSIP endpoint's AOR with the permanent contact
is never qualified if it is added to realtime storage
after asterisk started.
This patch adds qualifying for the AOR's permanent contacts
on the first handling of this AOR.
ASTERISK-26319 #close
Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe
A recent change attempted to optimize startup by not updating contact
status. Instead, code responsible for qualifying contacts updates the
status as it becomes known. The code even accounts for contacts/AORs
that are not set to be qualified.
The problem, though, is when there are no contacts associated with an
endpoint. A common case is when an endpoint is set to register its
contacts but has not done so yet. In this case, prior to registration,
the endpoint's device state will appear to be "not in use" and hints
associated with that device will appear to be "idle". In actuality, the
device state and hint should both appear as "unavailable". The reason
for the failure is that the optimization change made all persistent
endpoint states set to "unknown".
The fix here is to change the hard-coded "unknown" to be "offline"
instead. The default state will be offline until the qualifying code
determines that the contact is actually online. This way, if there are
no contacts at all, then the state stays as offline, and device state
and hints appear correctly.
ASTERISK-26269 #close
Reported by nappsoft
Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a
We may check a global config option hundreds of times a second or more.
Asking sorcery for the global configuration from the config files backend
involves several allocations and container traversals. Using realtime
without a memory cache is a lot worse because you have to lookup in the
realtime database each time to reconstitute the sorcery object. With a
memory cache for realtime, there is about the same amount of overhead as
for config files. Either way, it is still fairly expensive to access the
sorcery object that much.
* Cache the global config options so we can access them faster. You must
now always perform a res_pjsip reload to change the global options.
Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7
ast_channel_get_t38_state() calls ast_channel_queryoption() with
AST_OPTION_T38_STATE. If the passed in channel is a local channel then a
deadlock can happen if a channel lock is held when called.
* Made ast_channel_get_t38_state() callers not hold a channel lock before
calling.
* Update ast_channel_get_t38_state() doxygen to note that no channel locks
can be held when calling the function.
ASTERISK-26203 #close
Reported by: Etienne Lessard
ASTERISK-24822 #close
Reported by: David Brillert
ASTERISK-22732 #close
Reported by: Richard Mudgett
Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214
ASTERISK-25980 added the FAXMODE channel variable to res_fax.c.
Unfortunately, it also introduced a deadlock potential because
set_channel_variables() which sets FAXMODE can be called during a
masquerade. The ast_channel_get_t38_state() which gets the value used to
set FAXMODE cannot be called with the channel locked. As a result, local
channels can deadlock because of how they must acquire the locks necessary
to operate.
The intent of FAXMODE is for dialplan to know how a fax was transferred
after the fax completes. However, the previous patch sets FAXMODE to the
channel's current T.38 state AFTER the fax has completed and where T.38
may have already disconnected.
* Set FAXMODE based upon T.38 negotiations exchanged either with the fax
applications or the fax framehooks.
ASTERISK-26203
Reported by: Etienne Lessard
ASTERISK-24822
Reported by: David Brillert
ASTERISK-22732
Reported by: Richard Mudgett
Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1
fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be
called with any channel locks already held. A deadlock can happen if the
function is operating on a local channel.
* Made fax_gateway_indicate_t38() unlock the channel before calling
ast_indicate_data() since fax_gateway_indicate_t38() is always called with
the channel locked.
* Made fax_gateway_indicate_t38() return void since nothing cared about
its return value.
ASTERISK-26203
Reported by: Etienne Lessard
ASTERISK-24822
Reported by: David Brillert
ASTERISK-22732
Reported by: Richard Mudgett
Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407
The compilation failed for devmode
--enable DONT_OPTIMIZE
--enable BETTER_BACKTRACES
--enable DO_CRASH
--enable TEST_FRAMEWORK
res_pjsip/pjsip_configuration.c: In function dtls_handler:
res_pjsip/pjsip_configuration.c:974:20: error:
back may be used uninitialized in this function [-Werror=maybe-uninitialized]
int size = strlen(front);
^
cc1: all warnings being treated as errors
Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580
When res_odbc_transaction depended on res_odbc, it got the generic_odbc
headers and libs implicitly. Now that it no longer depends on res_odbc,
its dependency on generic_odbc must be explicit.
Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911
updated the uri handler to include the url prefix of the http server
this enables res_ari to add it to the uris when generating docs
Change-Id: I279335a2625261a8492206c37219698f42591c2e
(cherry picked from commit 6f448f32fe)
Historically, Asterisk has always specified annexb=no for the g729 format.
However, when using res_pjsip no format attribute was specified. This patch
makes it so the SDP now contains a format attribute line with annexb=no.
Note, that this means only g729a is negotiated. Even for pass through support.
According to rfc7261 the type of annex used (a or b) is dependent upon the
answerer. However, Asterisk being a back to back user agent makes this tricky
to support at this time, thus we only allow annex 'a' for now.
ASTERISK-26228 #close
patches:
res_format_attr_g729.c submitted by Jason Parker (license 4993)
Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0
Given resource paths did not have 'json' substituted in for the '{format}'. For
some auto generated documentation/comment strings it resulted in something like
the following:
"... REST handler for /api-docs/sounds.{format}"
This patch makes sure the resource api's path is properly substituted.
ASTERISK-25472 #close
Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23
The MODULEINFO dependencies between these 2 modules was reversed.
res_odbc should depend on res_odbc_transaction, not the other way
around.
ASTERISK-25984 #close
Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f
contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
Commit 1b666549f3 broke the srv failover
functionality if a TCP connection gets disconnected. Under these
conditions, session_inv_on_state_changed() gets a
PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new
transport. Unfortunately, session_inv_on_tsx_state_changed() also gets
the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates
the session.
* Made session_inv_on_tsx_state_changed() complete terminating the session
on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still
PJSIP_INV_STATE_DISCONNECTED.
ASTERISK-26305 #close
Reported by: Richard Mudgett
Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d
This updates func_channel.c and main/message.c to use a generic xpointer
include instead of including info from each channel driver. Now the
name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
documentation for func_channel. Setting the name attribute of info to
MessageToInfo or MessageFromInfo causes it to be included in the
MessageSend application and AMI action.
Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea
The Exchanging Device and Mailbox States could not working
if the Entity ID (EID) is not set manually and can't be obtained
from ethernet interface.
This patch replaces debug message to warning
and addes missing description about option 'entityid' to
asterisk.conf.sample.
With this patch the asterisk also:
(1) decline loading the modules which won't work without EID:
res_corosync and res_pjsip_publish_asterisk.
(2) warn if EID is empty on loading next modules:
pbx_dundi, res_xmpp
Starting with v197 systemd/udev will automatically assign "predictable"
names for all local Ethernet interfaces.
This patch also addes some new ethernet prefixes "eno" and "ens".
ASTERISK-26164 #close
Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6
* Groups of AGI commands that have similar functionality now reference
each other, and all reference the AGI application for ease of wiki
reference.
* The documentation for the AGI application has been improved, in
particular noting the various AGI types and how they are invoked.
* A warning message has been added to DeadAGI, noting that it is
deprecated.
Change-Id: I479ccdee8a7393f01b18692c3d4ab7e6bdd1875d
When compact_headers was set, we were sending a zero-length header name
for PAI and RPID because we always forced the short header name length
to 0. We did this because we cloned the header from "From" and wanted
to clear "f" from the sname. By cloning however, we bypass pjproject's
automatic logic that sets sname to name if there's no compact form of
the header, which there isn't for PAI and RPID. So now we force sname
to be the same as name right after we set name.
res_pjsip_diversion needed the same treatment for the Diversion header.
ASTERISK-26241 #close
Change-Id: I633ec139630cd83809aae00336cee4a10077e467
If debug was specified in the global configuration but left blank,
the logger would treat it as a wildcard and log all hosts. If
default_from_user was empty, a crash would result.
The global apply handler now checks for empty strings.
ASTERISK-26239 #close
ASTERISK-26238 #close
Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336
* Eliminated RAII_VAR() usage in
ast_sip_persistent_endpoint_update_state().
* Added a missing allocation failure check to
persistent_endpoint_find_or_create().
* Made persistent_endpoint_find_or_create() create the new object without
a lock as it isn't needed.
* Cleaned up some ao2 container allocation idioms.
* Reordered res_pjsip_mwi.c load_module() and unload_module()
Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8
* Eliminated most RAII_VAR() usage.
* Added several missing allocation failure checks.
* Made ast_sip_for_each_contact() allocate the wrapper ao2 object without
a lock as it is not needed.
Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc
The named aor lock was always being locked for writes so a rwlock adds no
benefit and may be slower because rwlocks are biased toward read locking.
Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28
A patch made to the master branch (Now the 14 branch) inadvertently made
libsrtp a required dependency in order to compile Asterisk. Rather than
create dummy defines to substitute for the defines supplied by libsrtp
when libsrtp is not available, most of the code in sdp_srtp.c is moved
into res_srtp.c. This gets more code out of Asterisk's core that isn't
used when SRTP is not available. This also makes another inadvertent
required dependency on libsrtp by Asterisk's core unlikely.
ASTERISK-26253 #close
Reported by: Ben Merrills
Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7
If both channels which should be masqueraded
are in the same serializer:
1st channel will be locked waiting condition 'complete'
2nd channel will be locked waiting condition 'suspended'
On heavy load system a chance that both channels will be in
the same serializer 'pjsip/distibutor' is very high.
To reproduce compile res_pjsip/pjsip_distributor.c with
DISTRIBUTOR_POOL_SIZE=1
Steps to reproduce:
1. Party A calls Party B (bridged call 'AB')
2. Party B places Party A on hold
3. Party B calls Voicemail app (non-bridged call 'BV')
4. Party B attended transfers Party A to voicemail using REFER.
5. When asterisk masquerades calls 'AB' and 'BV',
a deadlock is happened.
This patch adds a suspension indicator to the taskprocessor.
When a session suspends/unsuspends the serializer
it sets the indicator to the appropriate state.
The session checks the suspension indicator before
suspend the serializer.
ASTERISK-26145 #close
Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b
libunbound at version 1.4.20 (which CentOS still uses) declared all
of their string function parameters as as 'char *'. 1.4.21 changed
them all to 'const char *'. Thankfully 1.4.21 also introduced the
UNBOUND_VERSION_MAJOR define so configure now checks for that and
sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks
that and casts away the 'const' if it's not set.
Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and
Fedora24 (1.5.4). There are a few failing tests to be addressed though.
ASTERISK-26283 #close
Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148
When an RTCP packet is sent or received, res_rtp_asterisk generates a
Stasis event that contains the RTCP report as well as the local and
remote addresses that the report pertains to.
The addresses are determined using ast_find_ourip(). For the local
address, this will typically result in a lookup of the hostname of the
server, and then a DNS lookup of that hostname. If you do not have the
host in /etc/hosts, then this results in a full DNS lookup, which can
potentially block for some time.
This is especially problematic when performing RTCP reads, since those
are done on the same thread responsible for reading and writing media.
This patch addresses the issue by performing a lookup of the local
address when RTCP is allocated. We then use this cached local address
for the Stasis events when necessary.
ASTERISK-26280 #close
Reported by Mark Michelson
Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556
The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.
This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.
This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.
ASTERISK-26230 #close
Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a
When is executed CLI command "odbc show all" every time is show
information about variable last_negative_connect. If not there a fail
attempt of connection will show date like "1969-12-31 21:00:00".
This patch fix there situation for to show only this information when
exists a fail attempt before.
Change-Id: I7c058b0be6f7642e922de75ee6b82c7276c9f113
This change replaces the custom unload process for the outbound
publish module with the common serializer shutdown group.
ASTERISK-25217 #close
Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6
This changes the use of an empty regex for both res_sorcery_config
and res_sorcery_memory to "." instead. This is a more compatible
regular expression which also works on FreeBSD.
ASTERISK-26206 #close
Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388
Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk
defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests
this.
ASTERISK-26218 #close
ASTERISK-26221 #close
Reported by: Aaron Meriwether
Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa
This changes context switches from a linked list to a vector, makes
'struct ast_sw' opaque to pbx.c.
Although ast_walk_context_switches is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_switches_count (AST_VECTOR_SIZE)
* ast_context_switches_get (AST_VECTOR_GET)
As with ast_walk_context_switches callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the switches, they have been converted to use the new functions.
Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998
This patch removed call of pjsip_tx_data_dec_ref in send_notify
if send_request failed.
The pjsip_dlg_send_request deletes the message on error by itself.
It seems this patch fixes next issues:
ASTERISK-26199
ASTERISK-26166
ASTERISK-26174
Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a
The fax_detect_framehook() has the potential to deadlock if an incoming
fax happens during the Playback or similar application.
* Fixed the potential deadlock by not calling ast_async_goto() with the
channel lock held.
* Made always eat the fax detection frame whether there is a fax extension
or not.
* Made only detach the framehook if we detected a fax and not on other
possible frames.
ASTERISK-26216
Reported by: Richard Mudgett
Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d
The fax detection timeout option did not work because basically the wrong
variable was checked in fax_detect_framehook(). As a result, the timer
would timeout immediately and disable fax detection.
* Fixed ignoring negative timeout values. We'd complain and then go right
on using the negative value.
* Fixed destroy_faxdetect() in the off-nominal case of an incomplete
object creation.
* Added more range checking to FAXOPT(gateway) timeout parameter.
ASTERISK-26214 #close
Reported by: Richard Mudgett
Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976
The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call. The new feature is disabled if the timeout is set
to zero. The option is disabled by default.
ASTERISK-26214
Reported by: Richard Mudgett
Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d
If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead
code. This places all existing unit tests into a conditional block if
they weren't already.
ASTERISK-26211 #close
Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686
The function create_mwi_subscriptions_for_endpoint checks
if there is active contacts by retrieving aors and contacts.
This function is used to create all unsolicited mwi subscriptions
on startup and is used when contact added.
In both cases it's not necessary to check if there are contacts.
The contacts are needed when asterisk sends mwi.
ASTERISK-26200 #close
Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa
With this change, the initial RTP sequence number is randomly chosen not between
0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over
counter (ROC) synchronization is not lost for sRTP, when the very first RTP
packets get lost; see http://srtp.sourceforge.net/faq.html#Q6
ASTERISK-26207 #close
Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464
This changes context ignore patterns from a linked list to a vector,
makes 'struct ast_ignorepat' opaque to pbx.c.
Although ast_walk_context_ignorepats is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_ignorepats_count (AST_VECTOR_SIZE)
* ast_context_ignorepats_get (AST_VECTOR_GET)
As with ast_walk_context_ignorepats callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the ignorepats, they have been converted to use the new functions.
Change-Id: I78f2157d275ef1b7d624b4ff7d770d38e5d7f20a
This changes context includes from a linked list to a vector, makes
'struct ast_include' opaque to pbx.c.
Although ast_walk_context_includes is maintained the procedure is no
longer efficient except for the first call (inc==NULL). This
functionality is replaced by two new functions implemented by vector
macros.
* ast_context_includes_count (AST_VECTOR_SIZE)
* ast_context_includes_get (AST_VECTOR_GET)
As with ast_walk_context_includes callers of these functions are
expected to have locked contexts. Only a few places in Asterisk walked
the includes, they have been converted to use the new functions.
const have been applied where possible to parameters for ast_include
functions.
Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60
This commit adds scaffolding in order to support the SILK audio format
on calls. Roughly, this is what is added:
* Cached silk formats. One for each possible sample rate.
* ast_codec structures for each possible sample rate.
* RTP payload mappings for "SILK".
In addition, this change overhauls the res_format_attr_silk file in the
following ways:
* The "samplerate" attribute is scrapped. That's native to the format.
* There are far more checks to ensure that attributes have been
allocated before attempting to reference them.
* We do not SDP fmtp lines for attributes set to 0.
These changes make way to be able to install a codec_silk module and
have it actually work. It also should allow for passthrough silk calls
in Asterisk.
Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
aor_observer_deleted() needs to operate on all contacts found for the
deleted AOR instead of only the first one found. This is really only a
problem if there is more than one contact for the AOR.
Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1
* Fix some whitespace in various routines.
* Rename i to iter in persistent_endpoint_update_state().
* Fix off-nominal copy/paste message wording in
persistent_endpoint_contact_deleted_observer()
Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8
Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS)
support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added
for DTLS. The source code from main/tcptls.c should have been re-used to ease
security audits. Therefore, this change rolls back the change from July 2015 and
re-uses the code from July 2014. This has the additional benefits to work under
CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well.
ASTERISK-25659 #close
Reported by: StefanEng86, urbaniak, pay123
Tested by: sarumjanuch, traud
patches:
res_rtp_asterisk.patch submitted by sarumjanuch
dtls_centos_step_1.patch submitted by traud
dtls_centos_step_2.patch submitted by traud
Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c
It is possible in a hypothetical situation for a session refresh to be
invoked on a PJSIP when the negotiatior on the INVITE session has not
yet been established. While this shouldn't occur with existing uses of
ast_sip_session_refresh, the crashes that occur due to improperly
calling PJSIP functions that expect a non-NULL negotiatior are
avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this
means that simply checking for the presence of the negotiator before
passing it to other PJSIP functions that use it is allowable. As such,
this patch adds checks for the presence of the negotiator before calling
PJSIP functions that assume it is non-NULL.
Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d
When something very sad and wrong occurs, it's challenging sometimes to
figure out why. This patch adds some additional debug statements on
off-nominal paths to try and make debugging easier.
Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640
When res_corosync detects that a node leaves or joins, it currently is
informed of this via Corosync callbacks. However, there are a few
limitations with the information presented:
(1) While we have information that Corosync is aware of - such as the
Corosync nodeid - that information is really only useful inside of
Corosync or res_corosync. There's no way to translate a Corosync
nodeid to some other internally useful unique identifier for the
Asterisk instance that just joined or left the cluster.
(2) While res_corosync is notified of the instance joining or leaving
the cluster, it has no mechanism to inform the Asterisk core or
other modules of this event. This limits the usefulness of res_corosync
as a heartbeat mechanism for other modules.
This patch addresses both issues.
First, it adds the notion of a cluster discovery message both within the
Stasis message bus, as well as the binary event messages that
res_corosync uses to transmit data back and forth within the cluster.
When Asterisk joins the cluster, it sends a discovery message to the other
nodes in the cluster, which correlates the Corosync nodeid along with
the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids
to Asterisk EIDs, such that it can map changes in cluster state with the
Asterisk instance that has that nodeid. Likewise, when an Asterisk
instance receives a discovery message from a node in the cluster, it now
sends its own discovery message back to the originating node with the
local Asterisk EID. This lets Asterisk instances within the cluster
build a complete picture of the other Asterisk instances within the
cluster.
Second, it publishes the discovery messages onto the Stasis message bus.
Said messages are published whenever a node joins or leaves the cluster.
Interested modules can subscribe for the ast_cluster_discovery_type()
message under the ast_system_topic() and be notified when changes in
cluster state occur.
Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465
* We weren't properly subscribing to the channel and it's originator
on create.
* We weren't doing a publish_dial after calling ast_call on dial.
* We weren't calling depart_bridge when a channel left the dial bridge.
The first 2 issues were causing events to not be generated and the third
was actually causing channels to not get properly destroyed when hung up.
Together these 3 issues were causing the new
rest_apichannels/create_dial_bridge tests to fail.
As a result of the fixes, the cdr state machine had to be slightly
tweaked to allow bridge leave events without asserting and the tests
themselves had to be updated to account for the channels now cleaning
themselves up.
Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8
The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
patch introduced several regressions when the newly created "Updated"
state goes out for each endpoint registration refresh.
1) It restarted any OPTIONS RTT ping cycle.
2) It would interfere with a currently active ping and throw off that
ping's resulting RTT calculation.
3) It cleared the RTT time each time the endpoint was refreshed.
4) The cleared RTT time was sent out as a statsd update each time.
5) It created two AMI events for each update.
* Revert the original patch and reimplement it. Now the current contact
status state is re-sent instead of the state being momentarily toggled
every time the endpoint refreshes its registration. The statsd events are
not created for the re-sent refresh because they are sent after every
OPTIONS ping.
ASTERISK-26160 #close
Reported by: Matt Jordan
Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1
The func_odbc module was modified to ensure that the
previous behavior of using a single database connection
was maintained. This was done by getting a single database
connection and holding on to it. With the new multiple
connection support in res_odbc this will actually starve
every other thread from getting access to the database as
it also maintains the previous behavior of having only
a single database connection.
This change disables the func_odbc specific behavior if
the res_odbc module is running with only a single database
connection active. The connection is only kept for the
duration of the request.
ASTERISK-26177 #close
Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f
If the SQL UPDATE statement changes nothing then SQLRowCount returns 0.
This value should be treated as success.
But the function sorcery_realtime_update treats it as failed.
This bug was found using stress tests on PJSIP.
If there are 2 consecutive SIP REGISTER requests with the same contact data
during 1 second then res_pjsip_registrar adds contact location on 1st request
and tries to update contact location on 2nd.
The update fails and res_pjsip_registrar even removes correct contact location.
The test "object_update_uncreated" was removed from test_sorcery_realtime.c
because it's now a valid situation.
This patch also adds missing debug of extra SQL parameter.
ASTERISK-26172 #close
Change-Id: I05a7f3051455336c9dda29efc229decf86071303
Some T.38 implementations may send another re-invite after the initial
one which adds additional negotiation details (such as the max bitrate).
Currently this will fail when passthrough is being done in chan_sip as we
do nothing if T.38 is already active.
Other handlers of T.38 inside of Asterisk (such as res_fax) handle this
scenario so this change adds support for it to chan_sip and res_pjsip_t38.
If a request to negotiate is received while T.38 is already enabled a
new re-INVITE is sent and negotiation is done again.
ASTERISK-26179 #close
Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c
When using TCP transport with chan_pjsip, the TCP_NODELAY
option value was allocated on the stack, then passed as a
pointer to the tcp transport configuration structure, and
later re-used on subsequently created sockets when it was
no longer valid. This patch changes the allocation to be
a static.
ASTERISK-26180 #close
Reported by: Scott Griepentrog
Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0
If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.
ASTERISK-25471 #close
Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514
When an answer SDP is invalid we were disconnecting the outgoing call and
sending two BYE requests. The first BYE was sent by PJPROJECT because of
the invalid SDP answer. The second BYE was sent by Asterisk because it
thought the canceled call was the result of the RFC5407 section 3.1.2 race
condition.
* Made not send the BYE on a canceled session if the SDP negotiation is
incomplete because PJPROJECT has already sent a BYE for the failed
negotiation.
ASTERISK-25772 #close
Reported by: Dmitriy Serov
Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836
When an incoming call defers SDP negotiation and then sends us an invalid
SDP in the ACK, we need to send a BYE to disconnect the call. In this
case SDP negotiation has failed and we don't have valid media streams
negotiated.
ASTERISK-25772
Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8
Registering the PJMEDIA error codes allows errors found when parsing an
incoming SDP to be easier to figure out.
"Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)"
is much easier to understand than "Unknown error 220030".
ASTERISK-25772
Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0
pjsip_inv_end_session() is documented as being able to return the
passed in tdata parameter set to NULL on success.
Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047
This change removes hardcoded SDP parsing and generation for
Siren7 and Siren14 from chan_sip and moves it to format attribute
modules so it can also be used by chan_pjsip.
With this the fmtp lines for both are added with the bitrate
information.
ASTERISK-26021
Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037
fax_v21_session_new created a session details object but only released
the allocation reference during error conditions. fax_session_new adds
it's own reference to details if needed so the caller is always
responsible for cleaning it's own reference.
ASTERISK-26141 #close
Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88
The patch removes updating all Endpoints' status on startup.
Instead, only non-qualified aors with static contact
and non-qualified non-expired contacts are retrieved from the realtime to
update the endpoint status to ONLINE.
The endpoint name was added to the contact object to simply find the endpoint
that created this contact.
The status of endpoints with qualified aors will be updated by 'qualify'
functions.
ASTERISK-26061 #close
Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df
gcc 6 caught a previously unidentified self-comparison in
ice_candidate_cmp. Fixed it and re-ordered the predicates for better
short-circuiting.
ASTERISK-26140 #close
Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7
A non-existent constraint was being referenced in the upgrade script.
This patch corrects the problem by removing the reference.
In addition, the head of the alembic branch referred to a non-existent
revision. This has been fixed by referring to the proper revision.
This patch fixes another realtime problem as well. Our Alembic scripts
store booleans as yes or no values. However, Sorcery tries to insert
"true" or "false" instead. This patch introduces a new boolean type that
translates to "yes" or "no" instead.
ASTERISK-26128 #close
Change-Id: I51574736a881189de695a824883a18d66a52dcef
zuul
Richard Mudgett
Joshua Colp
Matt Jordan
George Joseph
Alexei Gradinari
Corey Farrell
Jean Aunis
Mark Michelson
David M. Lee
Torrey Searle
Kevin Harwell
Rodrigo Ramírez Norambuena
Alexander Traud
Scott Griepentrog