Commit Graph

155 Commits

Author SHA1 Message Date
George Joseph b748038230 res_pjsip_pubsub: Check for Content-Type header in rx_notify_request
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
2017-09-19 13:52:17 -05:00
George Joseph 446d48fd49 res_pjsip: Add handling for incoming unsolicited MWI NOTIFY
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
2017-09-13 09:24:28 -05:00
George Joseph 65ed2ea311 res_pjsip_pubsub: Fix reference to released endpoint
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
2017-06-14 11:16:54 -05:00
George Joseph 747beb1ed1 modules: change module LOAD_FAILUREs to LOAD_DECLINES
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
2017-04-12 15:57:21 -06:00
George Joseph 5013d8f5d3 res_pjsip: Symmetric transports
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
2017-03-16 09:49:07 -06:00
George Joseph fb68db87b1 res_pjsip_pubsub: Remove unneeded endpoint unref
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
2017-02-27 20:09:36 -06:00
George Joseph 4bdf5d329f res_pjsip_pubsub: Correctly implement persisted subscriptions
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
2017-02-15 13:11:46 -06:00
Richard Mudgett 0ea3c371c5 res_pjsip_pubsub.c: Implement "pjsip show subscriptions" commands.
ASTERISK-23828 #close

Change-Id: Ifb8a3b61f447aedc58a8e6b36a810f7566018567
2017-01-23 18:07:04 -06:00
zuul 697fde39ac Merge "res_pjsip_pubsub.c: Fix incorrect message string wrapping." 2017-01-23 14:07:49 -06:00
Joshua Colp e2da0021b9 Merge "res_pjsip_pubsub.c: Fix AMI event list counts." 2017-01-23 11:10:25 -06:00
Richard Mudgett ef9164b9ca res_pjsip_pubsub.c: Fix AMI event list counts.
Fix the AMI PJSIPShowSubscriptionsInbound, PJSIPShowSubscriptionsOutbound,
and PJSIPShowResourceLists actions event counts.  The reported counts may
not necessarily be accurate depending on what happens.

The subscriptions count would be wrong if Asterisk ever has outbound
subscriptions.

The resource list count could be wrong if a list were added or removed
during the AMI action being processed.

Change-Id: I4344301827523fa174960a42c413fd19abe4aed5
2017-01-20 12:39:41 -06:00
Richard Mudgett ab858295a2 res_pjsip_pubsub.c: Fix incorrect message string wrapping.
Change-Id: Id771e6fe56d89ce365ddcbb423f820af97211120
2017-01-20 12:37:19 -06:00
Richard Mudgett 6d648185bc res_pjsip_pubsub.c: Eliminate trivial SCOPED_LOCK usage.
Change-Id: Ie0b69a830385452042fa19e7d267c6790ec6b6be
2017-01-20 12:33:56 -06:00
Richard Mudgett bb196323f9 res_pjsip: Fix tdata leaks in off nominal paths.
Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b
2016-11-10 17:15:59 -05:00
Richard Mudgett ba362822f3 res_pjsip: Add ignore_uri_user_options option.
This implements the chan_sip legacy_useroption_parsing option but with a
better name.

* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.

ASTERISK-26316 #close
Reported by: Kevin Harwell

Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
2016-09-09 17:13:02 -05:00
Alexei Gradinari 81ea024d93 res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice.
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
2016-07-21 11:29:15 -04:00
Matt Jordan c49833653b res/res_pjsip_pubsub: Add additional debug statements
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
2016-07-13 09:11:46 -05:00
George Joseph b57cd01404 res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription
Occasionally under load we'll attempt to send a final NOTIFY on a
subscription that's already been terminated and a SEGV will occur
down in pjproject's evsub_destroy function.  This is a result of a
race condition between all the paths that can generate a notify
and/or destroy the underlying pjproject evsub object:

 * The client can send a SUBSCRIBE with Expires: 0.
 * The client can send a SUBSCRIBE/refresh.
 * The subscription timer can expire.
 * An extension state can change.
 * An MWI event can be generated.
 * The pjproject transaction timer (timer_b) can expire.

Normally when our pubsub_on_evsub_state is called with a terminate,
we push a task to the serializer and return at which point the dialog
is unlocked.  This is usually not a problem because the task runs
immediately and locks the dialog again.  When the system is heavily
loaded though, there may be a delay between the unlock and relock
during which another event may occur such as the subscription timer
or timer_b expiring, an extension state change, etc.  These may also
cause a terminate to be processed and if so, we could cause pjproject
to try to destroy the evsub structure twice.  There's no way for us to
tell that the evsub was already destroyed and the evsub's group lock
can't tolerate this and SEGVs.

The remedy is twofold.

 * A patch has been submitted to Teluu and added to the bundled
   pjproject which adds add/decrement operations on evsub's group lock.

 * In res_pjsip_pubsub:
   * configure.ac and pjproject-bundled's configure.m4 were updated
     to check for the new evsub group lock APIs.
   * We now add a reference to the evsub group lock when we create
     the subscription and remove the reference when we clean up the
     subscription.  This prevents evsub from being destroyed before
     we're done with it.
   * A state has been added to the subscription tree structure so
     termination progress can be tracked through the asyncronous tasks.
   * The pubsub_on_evsub_state callback has been split so it's not doing
     double duty.  It now only handles the final cleanup of the
     subscription tree.  pubsub_on_rx_refresh now handles both client
     refreshes and client terminates.  It was always being called for
     both anyway.
   * The serialized_on_server_timeout task was removed since
     serialized_pubsub_on_rx_refresh was almost identical.
   * Missing state checks and ao2_cleanups were added.
   * Some debug levels were adjusted to make seeing only off-nominal
     things at level 1 and nominal or progress things at level 2+.

ASTERISK-26099 #close
Reported-by: Ross Beer.

Change-Id: I779d11802cf672a51392e62a74a1216596075ba1
2016-06-21 13:50:24 -05:00
Richard Mudgett 5b7b16a87f res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer.
* Resolves potential reentrancy problems if system restarted in the middle
of subscription message transactions.

* Fixes memory leak recreating persistent subscriptions when the
subscription resource tree could not be created.

ASTERISK-26088
Reported by:  Richard Mudgett

Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be
2016-06-09 10:32:06 -05:00
Richard Mudgett c2ae49249c res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions.
We must continue using the serializer that the original SUBSCRIBE came in
on for the dialog.  There may be retransmissions already enqueued in the
original serializer that can result in reentrancy and message sequencing
problems.  The "sip_transaction Unable to register SUBSCRIBE transaction
(key exists)" message is a notable symptom of this issue.

Outgoing subscriptions still create the pjsip/pubsub/<endpoint>
serializers for their dialogs.

ASTERISK-26088
Reported by:  Richard Mudgett

Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0
2016-06-09 10:32:06 -05:00
Joshua Colp d03e170ae7 res_pjsip_pubsub: Use common datastores container API.
This migrates res_pjsip_pubsub over to using the newly
introduce common datastores management API instead of using
its own implementations for both subscriptions and
publications.

As well the extension state data now provides a generic
datastores container instead of a subscription. This allows
the dialog-info+xml body generator to work for both
subscriptions and publications.

ASTERISK-25999 #close

Change-Id: I773f9e4f35092da0f653566736a8647e8cfebef1
2016-05-09 10:40:36 -03:00
Joshua Colp bc19d9a2b0 Merge "res_pjsip_exten_state: Check if body generator is available." 2016-04-29 14:33:01 -05:00
Joshua Colp d57847a7c7 Merge "res_pjsip_pubsub.c: Fix body generator registration race." 2016-04-29 13:33:43 -05:00
Richard Mudgett 0b5292525c res_pjsip_exten_state: Check if body generator is available.
When starting the extension state publishers, check if the requested
message body generator is available.  If not available give error message
and skip starting that publisher.

* res_pjsip_pubsub.c: Create new API if type/subtype generator
registered.

* res_pjsip_exten_state.c: Use new body generator API for validation.

ASTERISK-25922

Change-Id: I4ad69200666e3cc909d4619e3c81042d7f9db25c
2016-04-28 17:14:44 -05:00
Richard Mudgett 3af83ea2fb res_pjsip_pubsub.c: Add useful information to some messages.
Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a
2016-04-28 17:05:20 -05:00
Richard Mudgett 8e1b663b87 res_pjsip_pubsub.c: Fix body generator registration race.
Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67
2016-04-28 17:02:08 -05:00
George Joseph e2524fcee3 res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited
res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds
the Message-Account header to the MWI NOTIFY.  Also, specifying mailboxes
on endpoints for unsolicited mwi and on aors for subscriptions required
that the admin know in advance which the client wanted.  If you specified
mailboxes on the endpoint, subscriptions were rejected even if you also
specified mailboxes on the aor.

Voicemail extension:
* Added a global default_voicemail_extension which defaults to "".
* Added voicemail_extension to both endpoint and aor.
* Added ast_sip_subscription_get_dialog for support.
* Added ast_sip_subscription_get_sip_uri for support.

When an unsolicited NOTIFY is constructed, the From header is parsed, the
voicemail extension from the endpoint is substituted for the user, and the
result placed in the Message-Account field in the body.

When a subscribed NOTIFY is constructed, the subscription dialog local uri
is parsed, the voicemail_extension from the aor (looked up from the
subscription resource name) is substituted for the user, and the result
placed in the Message-Account field in the body.

If no voicemail extension was defined, the Message-Account field is not added
to the NOTIFY body.

mwi_subscribe_replaces_unsolicited:
* Added mwi_subscribe_replaces_unsolicited to endpoint.

The previous behavior was to reject a subscribe if a previous internal
subscription for unsolicited MWI was found for the mailbox.  That remains the
default.  However, if there are mailboxes also set on the aor and the client
subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal
subscription is removed and replaced with the external subscription.  This
allows an admin to configure mailboxes on both the endpoint and aor and allows
the client to select which to use.

ASTERISK-25865 #close
Reported-by: Ross Beer

Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea
2016-03-30 13:23:54 -05:00
George Joseph d2eb65f71e res_pjsip: Strip spaces from items parsed from comma-separated lists
Configurations like "aors = a, b, c" were either ignoring everything after "a"
or trying to look up " b".  Same for mailboxes,  ciphers, contacts and a few
others.

To fix, all the strsep(&copy, ",") calls have been wrapped in ast_strip.  To
facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were
updated to handle null pointers.

In some cases, an ast_strlen_zero() test was added to skip consecutive commas.

There was also an attempt to ast_free an ast_strdupa'd string in
ast_sip_for_each_aor which was causing a SEGV.  I removed it.

Although this issue was reported for realtime, the issue was in the res_pjsip
modules so all config mechanisms were affected.

ASTERISK-25829 #close
Reported-by: Mateusz Kowalski

Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2
2016-03-07 13:16:41 -06:00
George Joseph ba8adb4ce3 res_pjsip/config_transport: Allow reloading transports.
The 'reload' mechanism actually involves closing the underlying
socket and calling the appropriate udp, tcp or tls start functions
again.  Only outbound_registration, pubsub and session needed work
to reset the transport before sending requests to insure that the
pjsip transport didn't get pulled out from under them.

In my testing, no calls were dropped when a transport was changed
for any of the 3 transport types even if ip addresses or ports were
changed. To be on the safe side however, a new transport option was
added (allow_reload) which defaults to 'no'.  Unless it's explicitly
set to 'yes' for a transport, changes to that transport will be ignored
on a reload of res_pjsip.  This should preserve the current behavior.

Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf
2016-02-19 18:57:55 -06:00
Joshua Colp 5c400a0fed res_pjsip_pubsub: Move where the subscription is stored to after initialized.
A problem arose when testing the AMI subscription listing actions where it
was possible for a subscription that had not been fully initialized to be
listed. This was problematic as the underlying listing code would crash.

This change makes it so the subscription tree is fully set up before it is
added to the list of subscriptions. This ensures that when the listing actions
get the subscription it is valid.

ASTERISK-25738 #close

Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48
2016-02-15 13:01:54 -06:00
Mark Michelson b073244c51 res_pjsip_pubsub: Prevent crash from AMI command on freed subscription.
A test recently uncovered that running an ill-timed AMI command to show
inbound subscriptions could cause a crash since Asterisk will try to
operate on a freed subscription.

The fix for this is to remove the subscription tree from the list of
subscriptions at the time that we are sending our final NOTIFY request
out. This way, as the subscription is in the process of dying, it is
inaccessible from AMI.

Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23
2016-01-25 16:58:39 -06:00
Richard Mudgett 0bca2a5c26 res_pjsip: Create human friendly serializer names.
PJSIP name formats:
pjsip/aor/<aor>-<seq> -- registrar thread pool serializer
pjsip/default-<seq> -- default thread pool serializer
pjsip/messaging -- messaging thread pool serializer
pjsip/outreg/<registration>-<seq> -- outbound registration thread pool
serializer
pjsip/pubsub/<endpoint>-<seq> -- pubsub thread pool serializer
pjsip/refer/<endpoint>-<seq> -- REFER thread pool serializer
pjsip/session/<endpoint>-<seq> -- session thread pool serializer
pjsip/websocket-<seq> -- websocket thread pool serializer

Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084
2016-01-08 22:11:45 -06:00
Joshua Colp b522a5e30f res_pjsip_pubsub: Fix assertion when UAS dialog creation fails.
When compiled with assertions enabled one will occur when destroying
the subscription tree when UAS dialog creation fails. This is because
the code assumes that a dialog will always exist on a subscription
tree when in reality during this specific scenario it won't.

This change makes it so a dialog is not removed from the subscription
tree if it is not present.

ASTERISK-25505 #close

Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee
2015-10-29 08:33:08 -05:00
Mark Michelson 7f9823ff57 res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog.
A certain situation can result in our attempting to send a NOTIFY on a
destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but
that subscriber has dropped off the network. We end up retransmitting
that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY
transaction. When the pjsip evsub code is told that the transaction has
been terminated, it responds in kind by alerting us that the
subscription has been terminated, destroying the subscription, and then
removing its reference to the dialog, thus destroying the dialog.

The problem is that when we get told that the subscription is being
terminated, we detect that we have not sent a terminating NOTIFY
request, so we queue up such a NOTIFY to be sent out. By the time that
queued NOTIFY gets sent, the dialog has been destroyed, so attempting to
send that NOTIFY can result in a crash.

The fix being introduced here is actually a reintroduction of something
the pubsub code used to employ. We hold a reference to the dialog and
wait to decrement our reference to the dialog until our subscription
tree object is destroyed. This way, we can send messages on the dialog
even if the PJSIP evsub code wants to terminate earlier than we would
like.

In doing this, some NULL checks for subscription tree dialogs have been
removed since NULL dialogs are no longer actually possible.

Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5
2015-10-22 16:18:08 -05:00
Mark Michelson e9e4bc9ece res_pjsip_pubsub: Ensure dialog lock balance.
When sending a NOTIFY, we lock the dialog and then unlock the dialog
when finished. A recent change made it so that the subscription tree's
dialog pointer will be set NULL when sending the final NOTIFY request
out. This means that when we attempt to unlock the dialog, we pass a
NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog
remains locked after we think we have unlocked it. When a response to
the NOTIFY arrives, the monitor thread attempts to lock the dialog, but
it cannot because we never released the dialog lock. This results in
Asterisk being unable to process incoming SIP traffic any longer.

The fix in this patch is to use a local pointer to save off the pointer
value of the subscription tree's dialog when locking and unlocking the
dialog. This way, if the subscription tree's dialog pointer is NULLed
out, the local pointer will still have point to the proper place and the
dialog lock will be unlocked as we expect.

Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a
2015-10-22 16:18:08 -05:00
Mark Michelson b96267f7a3 res_pjsip_pubsub: Prevent crashes on final NOTIFY.
The SIP dialog is removed from the subscription tree when the final
NOTIFY is sent. However, after the final NOTIFY is sent, the persistence
update function still attempts to access the cseq from the dialog,
resulting in a crash.

This fix removes the subscription persistence at the same time that the
dialog is removed from the subscription tree. This way, there is no
attempt to update persistence when the subscription is being destroyed.

Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb
2015-10-22 16:18:08 -05:00
Mark Michelson 386cd7b2b0 res_pjsip_pubsub: Remove serializer when sending final NOTIFY.
There have been crashes seen where a taskprocessor's listener is NULL
unexpectedly.

Looking at backtraces, the problem was specifically seen in PJSIP
serializers.

Subscriptions make the mistake of removing a serializer from a dialog
during subscription tree destruction. Since subscription trees are
reference-counted, guaranteeing the circumstances behind the destruction
are not possible. This makes it so that the dialog serializer can be
removed while not holding the dialog lock. This makes it possible for
the distributor to get a pointer to the dialog serializer and have that
serializer get freed out from under it.

The fix for this is to remove the serializer from a subscription dialog
when sending the final NOTIFY. This guarantees that the serializer is
removed with the dialog lock held. By doing this, we guarantee that if
the distributor gains access to the dialog's serializer, it will not be
possible for the serializer to get freed by another thread.

Change-Id: I21f5dac33529f65cec45679bdace60670800ff66
2015-10-22 16:17:47 -05:00
Mark Michelson 0b63d011c9 res_pjsip_pubsub: Fix crash on destruction of empty subscription tree.
If an old persistent subscription is recreated but then immediately
destroyed because it is out of date, the subscription tree will have no
leaf subscriptions on it. This was resulting in a crash when attempting
to destroy the subscription tree.

A simple NULL check fixes this problem.

Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac
2015-10-22 15:39:58 -05:00
Mark Michelson ac0194dad6 res_pjsip_pubsub: Solidify lifetime and ownership of objects.
There have been crashes and general instability seen in the pubsub code,
so this patch introduces three changes to increase the stability.

First, the ownership model for subscriptions has been modified. Due to
RLS, subscriptions are stored in memory as a tree structure. Prior to my
patch, the PJSIP subscription was the owner of the subscription tree.
When the PJSIP subscription told us that it was terminating, we started
destroying the subscription tree along with all of the individual leaf
subscriptions that belong to the tree. The problem with this model is
that the two actors in play here, the PJSIP subscription and the
individual leaf subscriptions, need to have joint ownership of the
subscription tree. So now, the PJSIP subscription and the individual
leaf subscriptions each have a reference to the subscription tree. This
way, we will not actually free memory until no players are left that
care. The PJSIP subscription is a bigger stakeholder, in that if the
PJSIP subscription's reference to the subscription tree is removed, the
subscription tree instructs the leaf subscriptions to shut down and drop
their references to the subscription tree when possible. The individual
leaf subscriptions, upon being told to shut down, can drop their stasis
subscriptions or whatever they use to learn of new state, and then drop
their reference to the subscription tree once they are ready to die.

Second, the lifetime of a PJSIP subscription's reference to our
subscription tree has been altered. As I learned from doing a deep dive,
the PJSIP evsub code can tell Asterisk multiple times that the
subscription has been terminated, and not all of these times
are especially helpful. I have altered the message flow that we use for
SIP subscriptions such that we will always drop the PJSIP subscription's
reference to the subscription tree when we send the NOTIFY that
terminates a SIP subscription. This also means that we will now queue
NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so
that we can have predictable state changes from the PJSIP evsub code.

Third, the synchronization of operations has been improved. PJSIP can
call into our code from a serializer thread (e.g. upon receiving an
incoming request) or from the monitor thread (e.g. when a subscription
times out). Because of this, there is the possibility of competing
threads stepping on each other. PJSIP attempts to do some
synchronization on its own by always keeping the dialog lock held when
it calls into us. However, since we end up pushing tasks into the
serializer, the result was that serialized operations were not grabbing
the dialog lock and could, as a result, step on something that was being
attempted by a different thread. Now we ensure that serialized
operations grab the dialog lock, then check for extenuating
circumstances, then proceed with their operation if they can.

Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5
2015-10-22 15:39:58 -05:00
Mark Michelson e47396721f res_pjsip_pubsub: Eliminate race during initial NOTIFY.
There is a slim chance of a race condition occurring where two threads
can both attempt to manipulate the same area.

Thread A can be handling an incoming initial SUBSCRIBE request. Thread A
lets the specific subscription handler know that the subscription has
been established.

At this point, Thread B may detect a state change on the subscribed
resource and queue up a notification task on Thread C, the subscription
serializer thread.

Now Thread A attempts to generate the initial NOTIFY request to send to
the subscriber at the same time that Thread C attempts to generate a
state change NOTIFY request to send to the subscriber.

The result is that Threads A and C can step on the same memory area,
resulting in a crash. The crash has been observed as happening when
attempting to allocate more space to hold the body for the NOTIFY.

The solution presented here is to queue the subscription establishment
and initial NOTIFY generation onto the subscription serializer thread
(Thread C in the above scenario). This way, there is no way that a state
change notification can occur before the initial NOTIFY is sent, and if
there is a quick succession of NOTIFYs, we can guarantee that the two
NOTIFY requests will be sent in succession.

Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815
2015-09-17 11:12:22 -05:00
Richard Mudgett 69824fdfbf res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.
Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093
2015-09-10 13:16:12 -05:00
Richard Mudgett 2526659432 res_pjsip_pubsub.c: Add some notification comments.
Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20
2015-09-10 13:16:12 -05:00
Richard Mudgett 9b290dfe2f res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.
We should not try to send a SIP response message because we may be
restoring a persistent subscription where we are not responding to a SIP
request.

Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec
2015-09-10 13:16:12 -05:00
Richard Mudgett 73eb132012 res_pjsip_pubsub.c: Fix off-nominal memory leak.
Fix off-nominal visited vector leak in build_resource_tree().

Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c
2015-09-10 13:16:12 -05:00
Richard Mudgett 2b30fc2b2d res_pjsip_pubsub.c: Fix one byte buffer overrun error.
ast_sip_pubsub_register_body_generator() did not account for the null
terminator set by sprintf() in the allocated output buffer.

Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2
2015-09-10 13:16:12 -05:00
Richard Mudgett 08a182c8e6 res_pjsip_pubsub.c: Use ast_alloca() instead of alloca().
Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3
2015-09-10 13:16:12 -05:00
Richard Mudgett 61f30db877 res_pjsip_pubsub.c: Add missing error return in load_module().
Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc
2015-09-10 13:16:12 -05:00
Mark Michelson beb568e51c res_pjsip_pubsub: re-re-fix persistent subscription storage.
A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as
a means of writing an appropriate packet to persistent storage. While
this partially solved the issue, it had its own problems.
pjsip_msg_print will always add a Content-Length header to the message
it prints. Frequent restarts of Asterisk can result in persistent
subscriptions being written with five or more Content-Length headers. In
addition, sometimes some apparent corruption of individual headers could
be seen.

This aims to fix the problem by not running a parsed message through an
interpreter but rather by taking the raw message and saving it. The
logic for what to save is going to be different depending on whether a
SUBSCRIBE was received from the wire or if it was pulled from
persistence. When receiving a packet from the wire, when using a
streaming transport, the rdata->pkt_info.packet may contain multiple SIP
messages or fragments. However, the rdata->msg_info.msg_buf will always
contain the current SIP message to be processed. When pulling from
persistence, though, the rdata->msg_info.msg_buf will be NULL since no
transport actually handled the packet. However, since we know that we
will always ever pull one SIP message from persistence, we are free to
save directly from rdata->pkt_info.packet instead.

ASTERISK-25365 #close
Reported by Mark Michelson

Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b
2015-09-01 09:41:10 -05:00
Joshua Colp a408369bac res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced
When recreating a subscription it is possible for a freed sub_tree
to be referenced when the initial NOTIFY fails to be created.

Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788
2015-08-24 11:09:05 -05:00
Mark Michelson 382334cc06 res_pjsip_pubsub: More accurately persist packet.
The pjsip_rx_data structure has a pkt_info.packet field on it that is
the packet that was read from the transport. For datagram transports,
the packet read from the transport will correspond to the SIP message
that arrived. For streamed transports, however, it is possible to read
multiple SIP messages in one packet.

In a recent case, Asterisk crashed on a system where TCP was being used.
This is because at some point, a read from the TCP socket resulted in a
200 OK response as well as an incoming SUBSCRIBE request being stored in
rdata->pkt_info.packet. When the SUBSCRIBE was processed, the
combination 200 OK and SUBSCRIBE was saved in persistent storage. Later,
a restart of Asterisk resulted in the crash because the persistent
subscription recreation code ended up building the 200 OK response
instead of a SUBSCRIBE request, and we attempted to access
request-specific data.

The fix here is to use the pjsip_msg_print() function in order to
persist SUBSCRIBE requests. This way, rather than using the raw socket
data, we use the parsed SIP message that PJSIP has given us. If we
receive multiple SIP messages from a single read, we will be sure only
to save off the relevant SIP message. There also is a safeguard put in
place to make sure that if we do end up reconstructing a SIP response,
it will not cause a crash.

ASTERISK-25306 #close
Reported by Mark Michelson

Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2
2015-08-06 13:15:59 -05:00
Richard Mudgett 7cd99be534 PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error.
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
2015-07-06 16:15:12 -05:00
Matt Jordan f66c41e668 res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS
In addition to specifying lists of 'presence' and 'message-summary',
users can also create lists of type 'dialog'. These should be treated in
the same fashion as 'presence'.

Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e
2015-05-22 12:27:56 -05:00
Rodrigo Ramírez Norambuena eec010829a AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro.
Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723
2015-05-13 16:34:23 -05:00
Joshua Colp e33682cae2 res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination
The res_pjsip_exten_state module currently has a race condition between
processing the extension state callback from the PBX core and processing
the subscription shutdown callback from res_pjsip_pubsub. There is currently
no synchronization between the two. This can present a problem as while
the SIP subscription will remain valid the tree it points to may not.
This is in particular a problem as a task to send a NOTIFY may get queued
which will try to use the tree that may no longer be valid.

This change does the following to fix this problem:

1. All access to the subscription tree is done within the task that
sends the NOTIFY to ensure that no other thread is modifying or
destroying the tree. This task executes on the serializer for the
subscriptions.

2. A reference to the subscription serializer is kept to ensure it
remains valid for the lifetime of the extension state subscription.

3. The NOTIFY task has been changed so it will no longer attempt
to send a NOTIFY if the subscription has already been terminated.

ASTERISK-25057 #close
Reported by: Matt Jordan

Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643
2015-05-07 07:42:10 -05:00
Joshua Colp bfdc766bf6 Merge "res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs." 2015-04-22 05:29:18 -05:00
Mark Michelson 6331be0638 res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs.
When SUBSCRIBE dialogs were established, we never associated
the endpoint that created the subscription with the dialog
we end up creating. In most cases, this ended up not causing
any problems.

The actual bug that was observed was that when a device that
was behind NAT established a subscription with Asterisk, Asterisk
would end up sending in-dialog NOTIFY requests to the device's
private IP addres instead of the public address of the NAT router.

When Asterisk receives the initial SUBSCRIBE from the device,
res_pjsip_nat rewrites the contact to the public address on which the
SUBSCRIBE was received. This allows for the dialog to have its target
address set to the proper public address. Asterisk then would send a 200
OK response to the SUBSCRIBE, then a NOTIFY with the initial
subscription state. The device would then send a 200 OK response to
Asterisk's NOTIFY.

Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat
did not rewrite the address in the Contact header. Then, when the PJSIP
dialog layer processed the 200 OK, PJSIP would perform a comparison
between the IP address in the Contact header and its saved target
address for the dialog. Since they differed, PJSIP would update the
target dialog address to be the address in the Contact header. From this
point, if Asterisk needed to send a NOTIFY to the device, the result was
that the NOTIFY would be sent to the private address that the device
placed in the Contact header.

The reason why res_pjsip_nat did not rewrite the address when it
received the 200 OK response was that it could not associate the
incoming response with a configured endpoint. This is because on a
response, the only way to associate the response to an endpoint is by
finding the dialog that the response is associated with and then finding
the endpoint that is associated with that dialog. We do not perform
endpoint lookups on responses. res_pjsip_pubsub skipped the step of
associating the endpoint with the dialog we created, so res_pjsip_nat
could not find the associated endpoint and therefore couldn't rewrite
the contact.

This commit message is like 50x longer than the actual fix.

ASTERISK 24981 #close
Reported by Mark Michelson

Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd
2015-04-21 05:01:58 -05:00
Scott Griepentrog 664d3263e4 res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced
This change makes the send_notify of the sub_tree
not happen when the sub_tree has been deleted due
to the notify call failing, which avoids a crash.

ASTERISK-24970 #close

Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf
2015-04-16 13:52:10 -05:00
Corey Farrell d7fc85e69d res_pjsip: Enable unload of all modules at shutdown.
* Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes
  caused by running PJSIP functions from non-PJSIP threads.
* Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing
  crashes in some cases.  In theory pj_shutdown() should take care of this.
* Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at
  shutdown.
* Resolve leaked config global in res_pjsip_notify.
* Unregister pubsub pjsip service module.
* Implement cleanup for res_pjsip_session.

ASTERISK-24731 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4498/
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-26 17:47:42 +00:00
Richard Mudgett e2d3215b83 HTTP: Stop accepting requests on final system shutdown.
There are three CLI commands to stop and restart Asterisk each.

1) core stop/restart now - Hangup all calls and stop or restart Asterisk.
New channels are prevented while the shutdown request is pending.

2) core stop/restart gracefully - Stop or restart Asterisk when there are
no calls remaining in the system.  New channels are prevented while the
shutdown request is pending.

3) core stop/restart when convenient - Stop or restart Asterisk when there
are no calls in the system.  New calls are not prevented while the
shutdown request is pending.

ARI has made stopping/restarting Asterisk more problematic.  While a
shutdown request is pending it is desirable to continue to process ARI
HTTP requests for current calls.  To handle the current calls while a
shutdown request is pending, a new committed to shutdown phase is needed
so ARI applications can deal with the calls until the system is fully
committed to shutdown.

* Added a new shutdown committed phase so ARI applications can deal with
calls until the final committed to shutdown phase is reached.

* Made refuse new HTTP requests when the system has reached the final
system shutdown phase.  Starting anything while the system is actively
releasing resources and unloading modules is not a good thing.

* Split the bridging framework shutdown to not cleanup the global bridging
containers when shutting down in a hurry.  This is similar to how other
modules prevent crashes on rapid system shutdown.

* Moved ast_begin_shutdown(), ast_cancel_shutdown(), and
ast_shutting_down().  You should not have to include channel.h just to
access these system functions.

ASTERISK-24752 #close
Reported by: Matthew Jordan

Review: https://reviewboard.asterisk.org/r/4399/
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
Kevin Harwell ca02121ef7 Investigate and fix memory leaks in Asterisk
Fixed memory leaks that were found in Asterisk.

ASTERISK-24693 #close
Reported by:  Kevin Harwell
Review: https://reviewboard.asterisk.org/r/4347/
........

Merged revisions 430999 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23 15:21:56 +00:00
Richard Mudgett c7ea108e02 Revert -r430452 It needs to be redone for the next major AMI version change instead.
ASTERISK-24049


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-12 18:09:27 +00:00
Richard Mudgett ef34a05f21 AMI: Remove no longer used parameter from astman_send_listack().
Follow-up issue to -r430435 from reviewboard review.

ASTERISK-24049
Review: https://reviewboard.asterisk.org/r/4315/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:53:49 +00:00
Richard Mudgett 52a7cdb101 AMI: Make AMI actions that generate event lists consistent.
* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList

* Incremented the AMI version to 2.7.0.

* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start".  The corresponding complete event always used "Complete".

* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.

* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots().  Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.

* Fixed minor protocol error in action_getconfig() when no requested
categories are found.  Each line needs to be formatted as "Header: text".

* Fixed off-nominal memory leak in manager_build_parked_call_string().

* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().

ASTERISK-24049 #close
Reported by: Jonathan Rose

Review: https://reviewboard.asterisk.org/r/4315/
........

Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
George Joseph 8786fe13a4 res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown
If you do a 'core (shutdown|restart) graceful' persistent subscriptions won't 
survive.  If you do a 'core (shutdown|restart) now' or asterisk terminates for 
some reason, they do.  Here's why...

When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to 
subscribers for each subscription.  This not only tells the subscribers that the 
dialog/state machine is done, it also frees the last reference to the 
subscription tree which causes the persistent subscription to get deleted from 
astdb.  When asterisk restarts, nothing's left.  Just preventing the delete from 
astdb doesn't work because we already told the subscriber to terminate the 
dialog so we can't restart it even if it was still in astdb.  Everything works 
OK if asterisk terminates unexpectedly because we never send the 'terminated' 
message so on restart, the subscription is still in astdb and the subscriber is 
none the wiser.

This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for 
persistent connections.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4318/
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-08 21:41:02 +00:00
Mark Michelson 53e5b377a0 Activate persistent subscriptions when they are recreated.
Prior to this change, recreating persistent subscriptions would
create the subscription but would not activate it. This led to subscriptions
being listed in the "NULL" state by diagnostics and not sending NOTIFYs
when expected.

Review: https://reviewboard.asterisk.org/r/4261
........

Merged revisions 429571 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-15 15:48:47 +00:00
Matthew Jordan 1106e8fd0f main/stasis: Allow subscriptions to use a threadpool for message delivery
Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.

For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
  a single message - the subscription is created, a message is published, the
  delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.

This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.

Review: https://reviewboard.asterisk.org/r/4193

ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-01 17:59:21 +00:00
Mark Michelson 2454505d5a Fix race condition where duplicated requests may be handled by multiple threads.
This is the Asterisk 13 version of the patch. The main difference is in the pubsub
code since it was completely refactored between Asterisk 12 and 13.

Review: https://reviewboard.asterisk.org/r/4175
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-14 14:40:17 +00:00
Kinsey Moore 86a4ce4957 PJSIP: Enforce module load dependencies
This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub
have loaded properly before attempting to load any modules that depend
on them since the module loader system is not currently capable of
resolving module dependencies on its own.

ASTERISK-24312 #close
Reported by: Dafi Ni
Review: https://reviewboard.asterisk.org/r/4062/
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2014-10-16 16:32:25 +00:00
Matthew Jordan 523da7d1b3 res/res_pjsip_pubsub: Fix typo in WARNING message
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2014-10-07 14:09:47 +00:00
Matthew Jordan f27f41a288 res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions
A subscription that has been persisted can - for various reasons - fail to be
re-created on startup. This patch resolves a number of crashes that occurred
when a subscription cannot be re-created on several off-nominal paths.

#SIPit31

ASTERISK-24368 #close
Reported by: Matt Jordan
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2014-10-06 00:13:58 +00:00
Richard Mudgett 270932635d Simplify UUID generation in several places.
Replace code using ast_uuid_generate() with simpler and faster code using
ast_uuid_generate_str().  The new code avoids a malloc(), free(), and
copy.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-29 21:18:54 +00:00
Mark Michelson de72f3edbc Add subscription state test events.
These are needed for a set of batched notification RLS tests that are
about to be committed to the testsuite.

Review: https://reviewboard.asterisk.org/r/3967
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2014-09-18 18:56:54 +00:00
Mark Michelson 79eac1ffca res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies.
res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE
arrives.
* It checks that there is a subscription handler for the Event
* It checks that there are body generators for the types in the Accept header

The problem is, there's nothing that ensures that these two things will
actually mesh with each other. For instance, Asterisk will accept a subscription
to MWI that accepts pidf+xml bodies. That doesn't make sense.

With this commit, we add some type information to the mix. Subscription
handlers state they generate data of type X, and body generators state
that they consume data of type X. This way, Asterisk doesn't end up in
some hilariously mismatched situation like the one in the previous paragraph.

ASTERISK-24136 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/3877
Review: https://reviewboard.asterisk.org/r/3878
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2014-09-18 16:09:25 +00:00
Mark Michelson c6bc44f700 Pre-allocate transmission data buffer for RLS NOTIFY requests.
PJSIP, unless a constant is modified at compilation time, limits
SIP requests to 4000 bytes. Full-state RLS notifications can easily
exceed this limit with moderately small lists.

This changeset allows for Asterisk to work around this size limit by
performing its own allocation of the transmission data buffer. This
way, Asterisk can allocate a buffer that exceeds the built-in maximum.

We still impose our own limit of 64000 bytes, mainly because making
allocations larger than that is a bit absurd.

ASTERISK-24181 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/3977
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2014-09-08 17:35:02 +00:00
Jonathan Rose ef5f7a0e32 res_pjsip_pubsub: Check supported headers for eventlist when subscribing to
resource list

https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan
According to the off-nominal plan, if evenlist support is not specified in a
SUBSCRIBE's supported header(s), that subscription should be rejected with an
error.

ASTERISK-23871
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3960/diff/#index_header
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-08 15:58:24 +00:00
Mark Michelson 327d67270f Fix bug that did not allow for multiple batched RLS notifications to be sent.
A misunderstanding of how the scheduler worked caused further batched notifications
beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after
the batched notification is sent. This way, further notifications can be scheduled
when they arise.
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2014-08-28 15:50:41 +00:00
Mark Michelson 49f8bd4ad4 Set the role for inbound subscriptions correctly.
This was causing the AMI show_subscriptions test in
the testsuite to fail since all subscriptions were being
seen as subscribers instead of notifiers.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-20 20:41:04 +00:00
Matthew Jordan ba5d5da60b Improve call forwarding reporting, especially with regards to ARI.
This patch addresses a few issues:

1) The order of Dial events have been changed when performing a call forward.
   The order has now been altered to
    1) Dial begins dialing channel A.
    2) When A forwards the call to B, we issue the dial end event to channel
       A, indicating the dial is being canceled due to a forward to B.
    3) When the call to channel B occurs, we then issue a new dial begin to
       channel B.

2) Call forwards are now reported on the calling channel, not the peer channel.

3) AMI DialEnd events have been altered to display the extension the call is
   being forwarded to when relevant.

4) You can now get the values of channel variables for channels that are not
   currently in the Stasis application. This brings the retrieval of channel
   variables more in line with the rest of channel read operations since they
   may be performed on channels not in Stasis.

ASTERISK-24134 #close
Reported by Matt Jordan

ASTERISK-24138 #close
Reported by Matt Jordan

Patches:
	forward-shenanigans.diff uploaded by Matt Jordan (License #6283)

Review: https://reviewboard.asterisk.org/r/3899
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2014-08-18 00:57:01 +00:00
Mark Michelson db0a97f8ce Fix crashing unit tests with regards to RLS.
The unit tests require a sorcery.conf file that has been
set up to store resource lists in memory rather than retrieving
from configuration.

With a setup that is not conducive to running the tests, a fault
in sorcery currently causes Asterisk to crash when attempting to
run any of the tests.

To get around the crash, this adds a function that verifies the
current environment and marks the tests as "not run" if the setup
is not correct.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-11 17:40:07 +00:00
Mark Michelson b4e33c81e3 Fix crash encountered by the testsuite.
Running testsuite tests locally produced no errors, but when
run using the continuous integration framework, crashes occurred.

The crashes occurred due to a refcounting error that had been fixed
for a similar situation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-11 16:03:41 +00:00
Kinsey Moore 965ba7c36f Fix build in dev mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07 20:11:15 +00:00
Mark Michelson 99d0bccd35 Add support for RFC 4662 resource list subscriptions.
This commit adds the ability for a user to configure
a resource list in pjsip.conf. Subscribing to this
list simultaneously subscribes the subscriber to all
resources listed. This has the potential to reduce
the amount of SIP traffic when loads of subscribers
on a system attempt to subscribe to each others' states.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07 19:26:32 +00:00
Joshua Colp a8829490b6 res_pjsip_publish_asterisk: Add support for exchanging device and mailbox state using SIP.
This module uses the inbound and outbound PUBLISH support to exchange device and mailbox
state between Asterisk instances. Each instance is configured to publish to the other and
requires no intermediary server. The functionality provided is similar to the XMPP and
Corosync support.

Review: https://reviewboard.asterisk.org/r/3780/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07 14:37:26 +00:00
Mark Michelson dcf1ad14da Add module support level to ast_module_info structure. Print it in CLI "module show" .
ASTERISK-23919 #close
Reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/3802



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25 16:47:17 +00:00
Scott Griepentrog 0a99e4099b astobj2: assert on invalid ref and backtrace cleanup
If a reference count goes negative, instead of
just logging that fact, be more helpful with a
backtrace and an assert that will DO_CRASH.

This patch also removes the duplicate ao2_bt()
function and cleans up extraneous usage of the
ast_log_backtrace() call.

Review: https://reviewboard.asterisk.org/r/3765/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-18 17:55:38 +00:00
Mark Michelson 688bb204dc Reverse logic during subscription persistence recreation.
In the abstraction effort, this bit of logic got messed up. We
want to recreate the persistence if things go well, not if things
fail.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-30 18:39:56 +00:00
Matthew Jordan 15dcaeef82 res_pjsip: Add ActionID to events created as a result of PJSIP AMI actions
A number of various PJSIP AMI actions were failing to parse out and place the
ActionID into their responses. This patch updates the various PJSIP actions
such that the passed in ActionID is emitted on any event list complete events,
as well as any intermediate events created as a result of the action.

#ASTERISK-23947 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/3675/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-27 13:50:02 +00:00
Mark Michelson bc8c08c609 Abstract PJSIP-specific elements from the pubsub API.
This helps to pave the way for RLS work that is to come.
Since this is a self-contained change and subscription
tests still pass, this work is being committed directly
to trunk instead of a working branch.

ASTERISK-23865 #close
Review: https://reviewboard.asterisk.org/r/3628



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-25 20:57:28 +00:00
Kinsey Moore 27430374db Fix build in devmode for GCC 4.10
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2014-06-12 20:17:37 +00:00
Kevin Harwell 870394c051 res_pjsip_pubsub: unauthenticated remote crash in PJSIP pub/sub framework
A remotely exploitable crash vulnerability exists in the PJSIP channel driver's
pub/sub framework. If an attempt is made to unsubscribe when not currently
subscribed and the endpoint's "sub_min_expiry" is set to zero, Asterisk tries
to create an expiration timer with zero seconds, which is not allowed, so an
assertion raised.

The fix was to reject a subscription that is attempting to unsubscribe when not
being already subscribed.  Asterisk now checks for this situation appropriately
and responds with a 400 instead of crashing.

AST-2014-005

ASTERISK-23489 #close
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2014-06-12 14:39:29 +00:00
Joshua Colp 58f4c18ab6 res_pjsip_pubsub: Persist subscriptions in sorcery so they are recreated on startup.
This change makes res_pjsip_pubsub persist inbound subscriptions in sorcery. By default
this uses the local astdb but it can also be configured to store within an outside
database. When Asterisk is started these subscriptions are recreated if they have not
expired. Notifications are sent to the devices which have subscribed and they are none
the wiser that the system has restarted.

Review: https://reviewboard.asterisk.org/r/3598/
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2014-06-12 11:34:36 +00:00
Kinsey Moore abd3e4040b Allow Asterisk to compile under GCC 4.10
This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09 22:49:26 +00:00
Kevin Harwell 6905ac0f5e res_pjsip_pubsub: Set the body generation result to 0 for a valid path
The result of the "ast_sip_pubsub_generate_body_content" was not
set/initialized.  Consequently, the nominal path potentially returned
an invalid value, thus not sending mwi notifications.
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2014-04-10 21:10:46 +00:00
Kinsey Moore fcb04d889a PJSIP: Ensure test event has new state
The change that fixed the pubsub test event's use of a dangling pointer
also changed when it was processed relative to the pjsip subscription
state change processing. This change corrects the order of events while
holding a reference to the pointer that was previously dangling.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-07 20:41:05 +00:00
Kinsey Moore 5d9a1281ee PJSIP: Fix crash introduced in r411671
The test event introduced in revision 411671 uses a dangling pointer to
access information about pubsub state changes. This moves the event to
within the lifetime of the pointer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-07 13:30:25 +00:00
Kinsey Moore 045285f8e3 res_pjsip_pubsub: Add test event for state change
This adds a test event when subscription state changes so that
integration tests may trigger new actions at the appropriate times.

Review: https://reviewboard.asterisk.org/r/3383/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-03 12:06:37 +00:00
Mark Michelson 584f9bafa0 Remove all PJSIP MWI-specific use from our MWI code.
PJSIP has built-in MWI code that could be useful to some
degree, but our utilization of the API actually made our
code a bit more cluttered since we had to have special
cases peppered throughout.

With this change, we move to using the pjsip_evsub API
instead, which streamlines the code by removing special
cases.

Review: https://reviewboard.asterisk.org/r/3205
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-13 18:52:08 +00:00
Mark Michelson f55abe9cf1 Decouple subscription handling from NOTIFY/PUBLISH body generation.
When the PJSIP pubsub framework was created, subscription handlers were required
to state what event they handled along with what body types they knew how to
generate. While this serves well when implementing a base RFC, it has problems
when trying to extend the body to support non-standard or proprietary body
elements. The code also was NOTIFY-specific, meaning that when the time comes
that we start writing code to send out PUBLISH requests with MWI or presence
bodies, we would likely find ourselves duplicating code that had previously been
written.

This changeset introduces the concept of body generators and body supplements. A
body generator is responsible for allocating a native structure for a given body
type, providing the primary body content, converting the native structure to a
string, and deallocating resources. A body supplement takes the primary body
content (the native structure, not a string) generated by the body generator and
adds nonstandard elements to the body. With these elements living in their own
module, it becomes easy to extend our support for body types and to re-use
resources when sending a PUBLISH request.

Body generators and body supplements register themselves with the pubsub core,
similar to how subscription and publish handlers had done. Now, subscription
handlers do not need to know what type of body content they generate, but they
still need to inform the pubsub core about what the default body type for a
given event package is. The pubsub core keeps track of what body generators and
body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core
will check that there is a subscription handler for the event in the SUBSCRIBE,
then it will check that there is a body generator that can provide the content
specified in the Accept header(s).

Because of the nature of body generators and supplements, it means
res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no
longer worry about body types, instead calling
ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body.

Review: https://reviewboard.asterisk.org/r/3150
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31 22:27:07 +00:00
Kevin Harwell 565198b44b res_pjsip_pubsub: potential crash on timeout
What seems to be happening is if a subscription has been terminated and the
subscription timeout/expires is less than the time it takes for all pending
transactions (currently on the subscription) to end then the subscription
timer will not have been canceled yet and sub will be null.  Since the
subscription has already been canceled nothing needs to be done so a null
check in the asterisk code is sufficient in working around this problem.

(closes issue ASTERISK-23129)
Reported by: Dan Jenkins
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28 23:40:28 +00:00
Joshua Colp c3d5c41dae res_pjsip_pubsub: Ensure dialog manipulation happens on proper thread.
When destroying a subscription we remove the serializer from its dialog
and decrease its reference count. Depending on which thread dropped the
subscription reference count to 0 it was possible for this to occur in
a thread where it is not possible.

(closes issue ASTERISK-22952)
Reported by: Matt Jordan
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2013-12-24 02:20:18 +00:00