This change allows TEL URI requests to come through for basic calls. The
allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To
headers will now allow TEL URIs, as well as the request URI.
Support is only for TEL URIs present in traffic from a remote party.
Asterisk does not generate any TEL URIs on its own.
ASTERISK-26894
Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a
Set termination state to old subscriptions to prevent queueing and sending
NOTIFY messages on exten/device state changes.
Postpone destruction of old subscriptions until all already queued tasks
that may be using old subscriptions have completed.
ASTERISK-29906
Change-Id: I96582aad3a26515ca73a8460ee6756f56f6ba23b
If there is scheduled notification, we must delete it
to avoid using destroyed subscriptions.
ASTERISK-29906
Change-Id: I1c644e5e15a8fe43eed8e4f9112f113cbf87a40f
Treat time_t's as entirely unique and use the POSIX API's for
converting to/from strings.
Lastly, a 64-bit integer formats as 20 digits at most in base10.
Don't need to have any 100 byte buffers to hold that.
ASTERISK-29674 #close
Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56
When asterisk generates the RLMI part of NOTIFY request,
the asterisk uses the local contact uri instead of the URI to which
the SUBSCRIBE request is sent.
Because of this mismatch some IP phones (for example Cisco 5XX) ignore
this list.
According
https://datatracker.ietf.org/doc/html/rfc4662#section-5.2
The first mandatory <list> attribute is "uri", which contains the uri
that corresponds to the list. Typically, this is the URI to which
the SUBSCRIBE request was sent.
https://datatracker.ietf.org/doc/html/rfc4662#section-5.3
The "uri" attribute identifies the resource to which the <resource>
element corresponds. Typically, this will be a SIP URI that, if
subscribed to, would return the state of the resource.
This patch makes asterisk to generate URI using SUBSCRIBE request URI.
ASTERISK-29961 #close
Change-Id: I1fcfc08fd589677f40608c59a4e143c45ee05f6c
This patch makes the Resource List Subscriptions (RLS) dynamic.
The asterisk updates the current subscriptions to reflect the changes
to the list on the subscriptions refresh. If list items are added,
removed, updated or do not exist anymore, the asterisk regenerates
the resource list.
ASTERISK-29906 #close
Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3
If Subscription refresh occurred between when the batched notification
was scheduled and the serialized notification was to be sent,
then new schedule notification task would never be added.
There are 2 threads:
thread #1. ast_sip_subscription_notify is called,
if notification_batch_interval then call schedule_notification.
1.1. The schedule_notification checks notify_sched_id > -1
not true, then
send_scheduled_notify = 1
notify_sched_id =
ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb....
1.2. The sched_cb pushes task serialized_send_notify to serializer
and returns 0 which means no reschedule.
1.3. The serialized_send_notify checks send_scheduled_notify if it's false
the just returns. BUT notify_sched_id is still set, so no more ast_sched_add.
thread #2. pubsub_on_rx_refresh is called
2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer
2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout
which calls send_notify
2.3. The send_notify set send_scheduled_notify = 0;
The serialized_send_notify should always unset notify_sched_id.
ASTERISK-29904 #close
Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875
Whereas BLFs allow to show a display name for each RLS entry,
the asterisk provides only the extension now.
This is not end user friendly.
This commit adds a new resource_list option, resource_display_name,
to indicate whether display name of resource or the resource name being
provided for RLS entries.
If this option is enabled, the Display Name will be provided.
This option is disabled by default to remain the previous behavior.
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
will be set as the Display Name.
The 'message-summary' is not supported yet.
ASTERISK-29891 #close
Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681
The last argument to ast_copy_string() is the buffer size, not the
number of characters, so we add 1 to avoid stamping out the final \n
in the persisted SUBSCRIBE message.
Change-Id: I019b78942836f57965299af15d173911fcead5b2
pjproject returns the dialog locked and with a reference. However,
in Asterisk the method that handles this decrements the reference
and removes the lock prior to returning. This makes it possible,
under some circumstances, for another thread to free said dialog
before the thread that created it attempts to use it again. Of
course when the thread that created it tries to use a freed dialog
a crash can occur.
This patch makes it so Asterisk now returns the newly created
dialog both locked, and with an added reference. This allows the
caller to de-reference, and unlock the dialog when it is safe to
do so.
In the case of a new SIP Invite the lock, and reference are now
held for the entirety of the new invite handling process.
Otherwise it's possible for the dialog, or its dependent objects,
like the transaction, to disappear. For example if there is a TCP
transport error.
ASTERISK-29057 #close
Change-Id: I5ef645a47829596f402cf383dc02c629c618969e
(cherry picked from commit 6baa4b53be)
When dealing with a lot of video streams on WebRTC
the resulting SDPs can grow to be quite large. This
effectively doubles the maximum size to allow more
streams to exist.
The res_http_websocket module has also been changed
to use a buffer on the session for reading in packets
to ensure that the stack space usage is not excessive.
Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01
This patch makes the usual necessary changes when upgrading to a new
version pjproject. For instance, version number bump, patches removed
from third-party, new *.md5 file added, etc..
This patch also includes a change to the Asterisk pjproject Makefile to
explicitly create the 'source/pjsip-apps/lib' directory. This directory
is no longer there by default so needs to be added so the Asterisk
malloc debug can be built.
This patch also includes some minor changes to Asterisk that were a result
of the upgrade. Specifically, there was a backward incompatibility change
made in 2.10 that modified the "expires header" variable field from a
signed to an unsigned value. This potentially effects comparison. Namely,
those check for a value less than zero. This patch modified a few locations
in the Asterisk code that may have been affected.
Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to
check a minimum version of pjproject at compile time.
ASTERISK-28899 #close
Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81
Each subscription needs to have a reference to the persisted data
for it, as well as the main JSON contained within the tree. When
recreating a subscription this did not occur and they both shared
the same reference.
ASTERISK-28714
Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a
Some body generators, such as dialog-info+xml, require storing state
information which is then conveyed in the NOTIFY request itself. Up
until now there was no way for such body generators to persist this
information.
Two new API calls have been added to allow body generators to set and
get persisted data. This data is persisted out alongside the normal
persistence information and allows the body generator to restore
state information or to simply use this for normal storage of state.
State is stored in the form of JSON and it is up to the body
generator to interpret this as needed.
The dialog-info+xml body generator has been updated to take advantage
of this to persist the version number.
ASTERISK-27759
Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de
There are some warning messages which are not informative without endpoint:
"No registered subscribe handler for event presence.winfo"
"No registered publish handler for event presence"
This patch adds an endpoint name to these messages.
Change-Id: Ia2811ec226d8a12659b4f9d4d224b48289650827
The following message:
"Subscription request from endpoint <blah> rejected. Expiration of 0 is invalid"
Would sometimes spam the log with warnings if Asterisk restarted and a bunch
of clients sent unsubscribes. This patch changes it from a warning to a debug
message.
Change-Id: I841ec42f65559f3135e037df0e55f89b6447a467
There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:
main/mwi.h
main/mwi.c
Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.
Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list. Remove ao2_container_alloc macro.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.
This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.
There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.
ASTERISK-28103
Change-Id: I99bee23895baa0a117985d51683f7963b77aa190
The alembic for the PJSIP subscription persistence table has the
"prune_on_boot" field as a boolean. While in Asterisk we are
tolerant of many different definitions of true and false in the
database we only accept "yes" and "no". This change makes the
field treated as a yes/no instead of an integer, thus storing
"yes" and "no" instead of "1" and "0".
Change-Id: Ic8b9211b36babefe78f70def6828a135a6ae7ab6
The function pubsub_on_rx_publish_request incorrectly uses
of AST_SCHED_REPLACE_UNREF.
The AST_SCHED_REPLACE_UNREF should unref old '_data'.
Because of this, there may be a double unref
of variable 'publication' when ast_sched_del is unsuccessful
that leads to use after free of the 'publication' in publish_expire.
ASTERISK-27956 #close
Change-Id: Ie0f0cfc7e036953d890b188656010b325a5cdc82
Incoming publications need to ensure that the module remains
loaded for the lifetime of them. This is now done by holding
a reference to the module while the publication exists. This
mirrors that of inbound subscriptions.
ASTERISK-27783
Change-Id: Ia98c95a15e11af25728d5fb3e56e12cda0cfc7c0
ast_sip_push_task_synchronous() did not necessarily execute the passed in
task under the specified serializer. If the current thread is any
registered pjsip thread then it would execute the task immediately instead
of under the specified serializer. Reentrancy issues could result if the
task does not execute with the right serializer.
The original reason ast_sip_push_task_synchronous() checked to see if the
current thread was a registered pjsip thread was because of a deadlock
with masquerades and the channel technology's fixup callback
(ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356)
involving call pickups avoided the original deadlock situation entirely.
The PJSIP channel technology's fixup callback no longer needed to call
ast_sip_push_task_synchronous().
However, there are a few places where this unexpected behavior is still
required to avoid deadlocks. The pjsip monitor thread executes callbacks
that do calls to ast_sip_push_task_synchronous() that would deadlock if
the task were actually pushed to the specified serializer. I ran into one
dealing with the pubsub subscriptions where an ao2 destructor called
ast_sip_push_task_synchronous().
* Split ast_sip_push_task_synchronous() into
ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer().
ast_sip_push_task_wait_servant() has the old behavior of
ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has
the new behavior where the task is always executed by the specified
serializer or a picked serializer if one is not passed in. Both functions
behave the same if the current thread is not a SIP servant.
* Redirected ast_sip_push_task_synchronous() to
ast_sip_push_task_wait_servant() to preserve API for released branches.
ASTERISK_26806
Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3
A deadlock can happen when the PJSIP monitor thread is shutting down a
connection oriented transport (TCP/TLS) used by a subscription at the same
time as another thread tries to send something for that subscription. The
deadlock is between the pjsip monitor thread attempting to get the dialog
lock and another thread sending something for that dialog when it tries to
get the transport manager lock.
* res_pjsip_pubsub.c: Avoid the deadlock by pushing the subscription
removal to the subscription serializer.
* res_pjsip_registrar.c: Pushed off incoming registration contact removals
to a default serializer as a precaution. Removing the contacts involves
sorcery access which in this case will involve database access. Depending
upon the setup, the database may not be on the same machine and could take
awhile. We don't want to hold up the pjsip monitor thread with
potentially long access times.
ASTERISK-27706
Change-Id: I56b647aea565f24dba33e9e5ebeed4cd3f31f8c4
When receiving a SUBSCRIBE request the Accept headers from it are
stored locally. This operation has a fixed limit of 32 Accept headers
but this limit was not enforced. As a result it was possible for
memory outside of the allocated space to get written to resulting
in a crash.
This change enforces the limit so only 32 Accept headers are
processed.
ASTERISK-27640
Reported By: Sandro Gauci
Change-Id: I99a814b10b554b13a6021ccf41111e5bc95e7301
In an earlier release, inbound registrations on a reliable transport
were pruned on Asterisk restart since the TCP connection would have
been torn down and become unusable when Asterisk stopped. This same
process is now also applied to inbound subscriptions.
Also fixed issues in res_pjsip_registrar where it wasn't handling the
monitoring correctly when multiple registrations came in over the same
transport.
To accomplish this, the pjsip_transport_event feature needed to
be refactored to allow multiple monitors (multiple subcriptions or
registrations from the same endpoint) to exist on the same transport.
Since this changed the API, any external modules that may have used the
transport monitor feature (highly unlikey) will need to be changed.
ASTERISK-27612
Reported by: Ross Beer
Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36
This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
Those SIP messages that create dialogs require a contact header to be present.
If the contact header was missing from the message it could cause Asterisk to
crash.
This patch checks to make sure SIP messages that create a dialog contain the
contact header. If the message does not and it is required Asterisk now returns
a "400 Missing Contact header" response. Also added NULL checks when retrieving
the contact header that were missing as a "just in case".
ASTERISK-27480 #close
Change-Id: I1810db87683fc637a9e3e1384a746037fec20afe
Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh
ASTERISK-24198 #close
Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
This change makes it so that any user of the pubsub
API that requests the remote URI receives only the URI.
Previously the entire string was returned, which could
contain a display name.
ASTERISK-27290
Change-Id: If1d0cd6630f0a264856d31d2a67933109187a017
When allocate_subscription fails to initialize fields of the new sub it
calls destroy_subscription.
Change-Id: I5b79c915ec216dc00c13c1e4172137864a4bec85
Prevent unload of the module as certain pjsip initialization functions
cannot be reversed. This required a reorder of the module_load so that
the non-reversable pjsip functions are not called until all potential
errors have been ruled out.
ASTERISK-24483
Change-Id: Iee900f20bdd6ee1bfe23efdec0d87765eadce8a7
pubsub_on_rx_notify_request wasn't checking for a null
Content-Type header before checking that it was
application/simple-message-summary.
ASTERISK-27279
Reported by: Ross Beer
Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52
A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to
receive unsolicited MWI NOTIFY requests and make them available to
other modules via the stasis message bus.
res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request"
that parses a simple-message-summary body and, if
endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state
with the voice-message counts from the message.
Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c
destroy_subscription was attempting to get the id of the
subscription tree's endpoint after we'd already called ao2_cleanup
on it causing a segfault.
Moved the cleanup until after the debug statement and since
endpoint could also be NULL at this point, check for that as well.
ASTERISK-27057 #close
Reported-by: Ryan Smith
Change-Id: Ice0a7727f560cf204d870a774c6df71e159b1678
In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
A new transport parameter 'symmetric_transport' has been added.
When a request from a dynamic contact comes in on a transport with
this option set to 'yes', the transport name will be saved and used
for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.
It's saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output. On the
outgoing request, if a transport wasn't explicitly set on the
endpoint AND the request URI is not a hostname, the saved transport
will be used and the 'x-ast-txp' parameter stripped from the
outgoing packet.
* config_transport was modified to accept and store the new parameter.
* config_transport/transport_apply was updated to store the transport
name in the pjsip_transport->info field using the pjsip_transport->pool
on UDP transports.
* A 'multihomed_on_rx_message' function was added to
pjsip_message_ip_updater that, for incoming requests, retrieves the
transport name from pjsip_transport->info and retrieves the transport.
If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter
containing the transport name is added to the incoming Contact header.
* An 'ast_sip_get_transport_name' function was added to res_pjsip.
It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a
transport name if endpoint->transport is set or if there's an
'x-ast-txp' parameter on the uri and the uri host is an ipv4 or
ipv6 address. Otherwise it returns NULL.
* An 'ast_sip_dlg_set_transport' function was added to res_pjsip
which takes an ast_sip_endpoint, a pjsip_dialog, and an optional
pjsip_tpselector. It calls ast_sip_get_transport_name() and if
a non-NULL is returned, sets the selector and sets the transport
on the dialog. If a selector was passed in, it's updated.
* res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas
were modified to call ast_sip_dlg_set_transport() instead of their
original logic.
* res_pjsip/create_out_of_dialog_request was modified to call
ast_sip_get_transport_name() and pjsip_tx_data_set_transport()
instead of its original logic.
* Existing transport logic was removed from endpt_send_request
since that can only be called after a create_out_of_dialog_request.
* res_pjsip/ast_sip_create_rdata was converted to a wrapper around
a new 'ast_sip_create_rdata_with_contact' function which allows
a contact_uri to be specified in addition to the existing
parameters. (See below)
* res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated
since all it did was transport selection and that is now done in
ast_sip_create_dialog_uac and ast_sip_create_dialog_uas.
* 'contact_uri' was added to subscription_persistence. This was
necessary because although the parsed rdata contact header has the
x-ast-txp parameter added (if appropriate),
subscription_persistence_update stores the raw packet which
doesn't have it. subscription_persistence_recreate was then
updated to call ast_sip_create_rdata_with_contact with the
persisted contact_uri so the recreated subscription has the
correct transport info to send the NOTIFYs.
* res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since
all it did was transport selection and that is now done in
ast_sip_create_dialog_uac.
* pjsip_message_ip_updater/multihomed_on_tx_message was updated
to remove all traces of the x-ast-txp parameter from the
outgoing headers.
NOTE: This change does NOT modify the behavior of permanent
contacts specified on an aor. To do so would require that the
permanent contact's contact uri be updated with the x-ast-txp
parameter and the aor sorcery object updated. If we need to
persue this, we need to think about cloning permanent contacts into
the same store as the dynamic ones on an aor load so they can be
updated without disturbing the originally configured value.
You CAN add the x-ast-txp parameter to a permanent contact's uri
but it would be much simpler to just set endpoint->transport.
Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f
When a subscription was being recreated and the endpoint wasn't
found, we were trying to unref the endpoint. This was causing
FRACKs. Removed the unref.
ASTERISK-26823 #close
Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164
This patch fixes 2 original issues and more that those 2 exposed.
* When we send a NOTIFY, and the client either doesn't respond or
responds with a non OK, pjproject only calls our
pubsub_on_evsub_state callback, no others. Since
pubsub_on_evsub_state (which does the sub_tree cleanup) does not
expect to be called back without the other callbacks being called
first, it just returns leaving the sub_tree orphaned. Now
pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE
which is what pjproject will set to tell us that it was the
transaction that timed out or failed and not the subscription
itself timing our or being terminated by the client. If is
TSX_STATE, pubsub_on_evsub_state now does the proper cleanup
regardless of the state of the subscription.
* When a client renews a subscription, we don't update the
persisted subscription with the new expires timestamp. This causes
subscription_persistence_recreate to prune the subscription if/when
asterisk restarts. Now, pubsub_on_rx_refresh calls
subscription_persistence_update to apply the new expires timestamp.
This exposed other issues however...
* When creating a dialog from rdata (which sub_persistence_recreate
does from the packet buffer) there must NOT be a tag on the To
header (which there will be when a client refreshes a
subscription). If there is one, pjsip_dlg_create_uas will fail.
To address this, subscription_persistence_update now accepts a flag
that indicates that the original packet buffer must not be updated.
New subscribes don't set the flag and renews do. This makes sure
that when the rdata is recreated on asterisk startup, it's done
from the original subscribe packet which won't have the tag on To.
* When creating a dialog from rdata, we were setting the dialog's
remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq.
When the client tried to resubscribe after a restart with the
correct cseq, we'd reject the request with an Invalid CSeq error.
* The acts of creating a dialog and evsub by themselves when
recreating a subscription does NOT restart pjproject's subscription
timer. The result was that even if we did correctly recreate the
subscription, we never removed it if the client happened to go away
or send a non-OK response to a NOTIFY. However, there is no
pjproject function exposed to just set the timer on an evsub that
wasn't created by an incoming subscribe request. To address this,
we create our own timer using ast_sip_schedule_task. This timer is
used only for re-establishing subscriptions after a restart.
An earlier approach was to add support for setting pjproject's
timer (via a pjproject patch) and while that patch is still included
here, we don't use that call at the moment.
While addressing these issues, additional debugging was added and
some existing messages made more useful. A few formatting changes
were also made to 'pjsip show scheduled tasks' to make displaying
the subscription timers a little more friendly.
ASTERISK-26696
ASTERISK-26756
Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e
Ben Ford
Alexei Gradinari
Philip Prindeville
Alexander Traud
Josh Soref
Sean Bright
Kevin Harwell
Joshua C. Colp
Kevin Harwell
Corey Farrell
Joshua Colp
Richard Mudgett
George Joseph
Jenkins2