asterisk/asterisk.git
9 months agopjproject_bundled: check whether UPDATE is supported on outgoing calls
Pirmin Walthert [Wed, 28 Nov 2018 07:14:12 +0000 (08:14 +0100)]
pjproject_bundled: check whether UPDATE is supported on outgoing calls

In ASTERISK-27095 an issue had been fixed because of which chan_pjsip was not
trying to send UPDATE messages when connected_line_method was set to invite.
However this only solved the issue for incoming INVITES. For outgoing INVITES
(important when transferring calls) the options variable needs to be updated
at a different place.

ASTERISK-28182 #close
Reported-by: nappsoft

Change-Id: I76cc06da4ca76ddd6dce814a8b97cc66b98aaf29

9 months agoMerge "Revert "app_voicemail: Remove need to subscribe to stasis""
George Joseph [Fri, 30 Nov 2018 13:30:35 +0000 (07:30 -0600)]
Merge "Revert "app_voicemail: Remove need to subscribe to stasis""

9 months agoMerge "bridges: Remove reliance on stasis caching"
George Joseph [Thu, 29 Nov 2018 21:05:33 +0000 (15:05 -0600)]
Merge "bridges:  Remove reliance on stasis caching"

9 months agoRevert "app_voicemail: Remove need to subscribe to stasis"
George Joseph [Thu, 29 Nov 2018 19:26:16 +0000 (12:26 -0700)]
Revert "app_voicemail: Remove need to subscribe to stasis"

This reverts commit 29115e23848cceee0e2763bc70e87cb311919cdd.

That commit closed a long standing hole which allowed subscriptions
to mailboxes that weren't configured in voicemail.conf.  This
caused an issue with FreePBX which depdended on that behavior.
The commit is being reverted until FreePBX can handle the new
behavior.

ASTERISK-28151
Reported by: Ronald Raikes

Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15

9 months agoMerge "jansson: Upgrade to 2.12."
Kevin Harwell [Thu, 29 Nov 2018 18:57:32 +0000 (12:57 -0600)]
Merge "jansson: Upgrade to 2.12."

9 months agotest_cel: Plug a few ref leaks
George Joseph [Mon, 26 Nov 2018 22:18:00 +0000 (15:18 -0700)]
test_cel:  Plug a few ref leaks

These are only a few of the leaks.  The large number of macros
and return paths in this file would make a weeks worth of work
to plug them all.

Change-Id: Ie2369fa944023d44767871c5c30974cb077ffb56

9 months agobridges: Remove reliance on stasis caching
George Joseph [Wed, 19 Sep 2018 19:34:41 +0000 (13:34 -0600)]
bridges:  Remove reliance on stasis caching

* The bridging core no longer uses the stasis cache for bridge
  snapshots.  The latest bridge snapshot is now stored on the
  ast_bridge structure itself.

* The following APIs are no longer available since the stasis cache
  is no longer used:
    ast_bridge_topic_cached()
    ast_bridge_topic_all_cached()

* A topic pool is now used for individual bridge topics.

* The ast_bridge_cache() function was removed since there's no
  longer a separate container of snapshots.

* A new function "ast_bridges()" was created to retrieve the
  container of all bridges.  Users formerly calling
  ast_bridge_cache() can use the new function to iterate over
  bridges and retrieve the latest snapshot directly from the
  bridge.

* The ast_bridge_snapshot_get_latest() function was renamed to
  ast_bridge_get_snapshot_by_uniqueid().

* A new function "ast_bridge_get_snapshot()" was created to retrieve
  the bridge snapshot directly from the bridge structure.

* The ast_bridge_topic_all() function now returns a normal topic
  not a cached one so you can't use stasis cache functions on it
  either.

* The ast_bridge_snapshot_type() stasis message now has the
  ast_bridge_snapshot_update structure as it's data.  It contains
  the last snapshot and the new one.

* cdr, cel, manager and ari have been updated to use the new
  arrangement.

Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369

9 months agoMerge "stasis: Segment channel snapshot to reduce creation cost."
Jenkins2 [Mon, 26 Nov 2018 20:07:47 +0000 (14:07 -0600)]
Merge "stasis: Segment channel snapshot to reduce creation cost."

9 months agoMerge "astobj2: Create function to copy weak proxied objects from container."
Joshua Colp [Mon, 26 Nov 2018 19:48:00 +0000 (13:48 -0600)]
Merge "astobj2: Create function to copy weak proxied objects from container."

9 months agoMerge "RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit"
Joshua Colp [Mon, 26 Nov 2018 19:47:32 +0000 (13:47 -0600)]
Merge "RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit"

9 months agostasis: Segment channel snapshot to reduce creation cost.
Joshua Colp [Wed, 7 Nov 2018 17:18:34 +0000 (13:18 -0400)]
stasis: Segment channel snapshot to reduce creation cost.

When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.

This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.

The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.

Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.

ASTERISK-28119

Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423

9 months agostasis: Use an implementation specific channel snapshot cache.
Joshua Colp [Wed, 10 Oct 2018 14:28:18 +0000 (11:28 -0300)]
stasis: Use an implementation specific channel snapshot cache.

Channels no longer use the Stasis cache for channel snapshots. Instead
they are stored in a hash table in stasis_channels which reduces the
number of Stasis messages created and allows better storage.

As a result the following APIs are no longer available since the stasis
cache is no longer used:
ast_channel_topic_cached()
ast_channel_topic_all_cached()

The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now return an ao2_container of ast_channel_snapshots rather than
a container of stasis_messages therefore you can't (and don't need
to) call stasis_cache functions on it.

The ast_channel_topic_all() function now returns a normal topic not
a cached one so you can't use stasis cache functions on it either.

The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data. It contains the
last snapshot and the new one.

ast_channel_snapshot_get_latest() still returns the latest snapshot.

The latest snapshot is now stored on the channel itself to eliminate
cache hits when Stasis messages that have the snapshot as a payload
are created.

ASTERISK-28102

Change-Id: I9334febff60a82d7c39703e49059fa3a68825786

9 months agoMerge "func_strings: HASHKEY - negative array index can cause corruption"
Joshua Colp [Mon, 26 Nov 2018 13:44:11 +0000 (07:44 -0600)]
Merge "func_strings: HASHKEY - negative array index can cause corruption"

9 months agojansson: Upgrade to 2.12.
Corey Farrell [Mon, 26 Nov 2018 12:09:11 +0000 (07:09 -0500)]
jansson: Upgrade to 2.12.

This brings in jansson-2.12, removes all patches that were merged
upstream.  README is created in third-party/jansson/patches to explain
how to add patches but also because the patches folder must exist for
the build process to succeed.

Change-Id: If0f2d541c50997690660c21fb7b03d625a5cdadd

9 months agoRTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit
Alexei Gradinari [Fri, 23 Nov 2018 15:40:50 +0000 (10:40 -0500)]
RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit

The marker bit set on the voice packet indicates the start
of a new stream and a new time stamp.
Need to reset the DTMF last sequence number and the timestamp
of the last END packet.

If the new time stamp is lower then the timestamp of the last DTMF END packet
the asterisk drops all DTMF frames as out of order.

This bug was caught using Cisco ip-phone SPA50X and codec g722.
On SIP session update the SPA50X resets stream indicating it with market bit
and a new timestamp is twice smaller then the previous.

ASTERISK-28162 #close

Change-Id: If9c5742158fa836ad549713a9814d46a5d2b1620

10 months agoastobj2: Remove legacy ao2_container_alloc routine.
Corey Farrell [Mon, 19 Nov 2018 20:10:02 +0000 (15:10 -0500)]
astobj2: Remove legacy ao2_container_alloc routine.

Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.  Remove ao2_container_alloc macro.

Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088

10 months agoastobj2: Create function to copy weak proxied objects from container.
Corey Farrell [Wed, 14 Nov 2018 11:02:20 +0000 (06:02 -0500)]
astobj2: Create function to copy weak proxied objects from container.

Create ao2_container_dup_weakproxy_objs to perform a similar function to
ao2_container_dup.  This function expects the source container to have
weakproxy objects, inserts the associated non-weak objects into the
destination container.  Orphaned weakproxy objects are ignored.

Create test for this new function and for ao2_weakproxy_find.

Change-Id: I898387f058057e08696fe9070f8cd94ef3a27482

10 months agoMerge "stasis: Remove stringfields and lock from change message."
Joshua Colp [Tue, 20 Nov 2018 11:06:01 +0000 (05:06 -0600)]
Merge "stasis: Remove stringfields and lock from change message."

10 months agoMerge "app_queue: Cleanup queue_ref / queue_unref routines."
Joshua Colp [Tue, 20 Nov 2018 11:05:32 +0000 (05:05 -0600)]
Merge "app_queue: Cleanup queue_ref / queue_unref routines."

10 months agofunc_strings: HASHKEY - negative array index can cause corruption
Kevin Harwell [Fri, 16 Nov 2018 20:45:23 +0000 (14:45 -0600)]
func_strings: HASHKEY - negative array index can cause corruption

This patch makes it so only matching non-empty key names, and keys created by
the HASH function are eligible for inclusion in the comma separated string. It
also fixes a bug where it was possible to write to a negative index if the
result buffer was empty.

ASTERISK-28159
patches:
  ASTERISK-28159.diff submitted by Michael Walton (license 6502)

Change-Id: I6e57fe7307dfd856271753aed5ba64c59b511487

10 months agoCI: Get job timeouts from environment
George Joseph [Mon, 19 Nov 2018 17:59:07 +0000 (10:59 -0700)]
CI: Get job timeouts from environment

The job timeouts were hard coded in the jenkinsfiles which
means changes had to go through gerrit.  Now they are taken
from the following environment variables (and their defaults) that
can be set in Jenkins configuration...

TIMEOUT_GATES =      "60 MINUTES"
TIMEOUT_DAILIES =    "3 HOURS"
TIMEOUT_REF_DEBUG =  "24 HOURS"
TIMEOUT_UNITTESTS =  "30 MINUTES"

Change-Id: I673a551c1780bf665a3bc160b245da574aa4bbab

10 months agoMerge "test_res_pjsip_scheduler: Fix possible write after free in scheduler_policy."
Joshua Colp [Mon, 19 Nov 2018 15:38:50 +0000 (09:38 -0600)]
Merge "test_res_pjsip_scheduler: Fix possible write after free in scheduler_policy."

10 months agoMerge "res/res_ari: Fix null endpoint handle"
Joshua Colp [Mon, 19 Nov 2018 15:37:22 +0000 (09:37 -0600)]
Merge "res/res_ari: Fix null endpoint handle"

10 months agoMerge "bridge_native_rtp.c: Fail native bridge if no framing match."
Joshua Colp [Mon, 19 Nov 2018 15:36:17 +0000 (09:36 -0600)]
Merge "bridge_native_rtp.c: Fail native bridge if no framing match."

10 months agoMerge "res_pjsip_caller_id: Use static pj_str_t for fromto header names."
Joshua Colp [Mon, 19 Nov 2018 14:40:05 +0000 (08:40 -0600)]
Merge "res_pjsip_caller_id: Use static pj_str_t for fromto header names."

10 months agoMerge "stasis: Add internal filtering of messages."
Joshua Colp [Mon, 19 Nov 2018 14:36:50 +0000 (08:36 -0600)]
Merge "stasis: Add internal filtering of messages."

10 months agoapp_queue: Cleanup queue_ref / queue_unref routines.
Corey Farrell [Mon, 19 Nov 2018 13:00:03 +0000 (08:00 -0500)]
app_queue: Cleanup queue_ref / queue_unref routines.

This replaces the inline functions with macros.  This removes the need
to directly use __ao2_ref, opts instead for standard ao2_bump and
ao2_cleanup macros.

Change-Id: If4e04e9bab2e3c883188437cb9f487b3e498a21b

10 months agobacktrace: Refactor ast_bt_get_symbols so it doesn't crash
George Joseph [Thu, 8 Nov 2018 15:53:44 +0000 (08:53 -0700)]
backtrace:  Refactor ast_bt_get_symbols so it doesn't crash

We've been seeing crashes in libbfd when we attempt to generate
a stack trace from multiple threads.  It turns out that libbfd
is NOT thread-safe.  It can cache the bfd structure and give it to
multiple threads without protecting itself.  To get around this,
we've added a global mutex around the bfd functions and also have
refactored the use of those functions to be more efficient and
to provide more information about inlined functions.

Also added a few more tests to test_pbx.c.  One just calls
ast_assert() and the other calls ast_log_backtrace().  Neither are
run by default.

WARNING:  This change necessitated changing the return value of
ast_bt_get_symbols() from an array of strings to a VECTOR of
strings.  However, the use of this function outside Asterisk is not
likely.

ASTERISK-28140

Change-Id: I79d02862ddaa2423a0809caa4b3b85c128131621

10 months agostasis: Remove stringfields and lock from change message.
Joshua C. Colp [Sun, 18 Nov 2018 23:53:14 +0000 (19:53 -0400)]
stasis: Remove stringfields and lock from change message.

When a subscribe or unsubscribe occurs a message is published
containing this information. This change makes it so that the
message no longer uses stringfields or a lock, as both are not
really needed for the message.

Change-Id: I3f4831931d79f94fd979baf48048738df5dc1632

10 months agopjsip: New function PJSIP_PARSE_URI to parse URI and return part of URI
Alexei Gradinari [Tue, 13 Nov 2018 15:28:28 +0000 (10:28 -0500)]
pjsip: New function PJSIP_PARSE_URI to parse URI and return part of URI

New dialplan function PJSIP_PARSE_URI added to parse an URI and return
a specified part of the URI.

This is useful when need to get part of the URI instead of cutting it
using a CUT function.

For example to get 'user' part of Remote URI
${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)}

ASTERISK-28144 #close

Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a

10 months agoMerge "pjproject-bundled: Use AST_DEVMODE for conditional compilation."
George Joseph [Sun, 18 Nov 2018 20:11:58 +0000 (14:11 -0600)]
Merge "pjproject-bundled: Use AST_DEVMODE for conditional compilation."

10 months agostasis: Add internal filtering of messages.
Joshua Colp [Sun, 23 Sep 2018 20:50:01 +0000 (17:50 -0300)]
stasis: Add internal filtering of messages.

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

10 months agoCI: Add tmpfs to all jenkinsfiles
George Joseph [Sun, 18 Nov 2018 16:38:40 +0000 (09:38 -0700)]
CI:  Add tmpfs to all jenkinsfiles

Change-Id: Ida29d70d48d5f39aabf0b25c66b51f79324a8cba

10 months agoMerge "res/res_pjsip_nat: Fix logic for REINVITES"
George Joseph [Sun, 18 Nov 2018 14:47:10 +0000 (08:47 -0600)]
Merge "res/res_pjsip_nat: Fix logic for REINVITES"

10 months agoMerge "taskprocessor: Prevent race creating new taskprocessor."
George Joseph [Sat, 17 Nov 2018 23:28:41 +0000 (17:28 -0600)]
Merge "taskprocessor: Prevent race creating new taskprocessor."

10 months agoCI: Mount a tmpfs on /tmp for testsuite docker containers
George Joseph [Sat, 17 Nov 2018 21:40:46 +0000 (14:40 -0700)]
CI:  Mount a tmpfs on /tmp for testsuite docker containers

Change-Id: I0566d81b0852f22066cd76d58eae5f1fda5602aa
(cherry picked from commit 73efe86436427e5f43c532e5d42505ab4ec104d9)

10 months agoCI: Pass work directory to runTestsuite
George Joseph [Sat, 17 Nov 2018 19:07:32 +0000 (12:07 -0700)]
CI:  Pass work directory to runTestsuite

The testsuite can now use a user-specified work directory for
all it's temp files.  This allows the docker containers to use
a tmpfs backed directory for the temp files instead of it's
own write-layer image.

* runTestsuite.sh now accepts a --work-dir command line argument
  that gets exported as AST_WORK_DIR before running the testsuite.

* gates.jenkinsfile now specifies --work-dir to be
  <testsuite_dir>/astroot.

Since the Asterisk CI docker hosts now mount /srv/jenkins/workspace
on a tmpfs, asterisk should be compiled and the testsuite run all in
memory.

Change-Id: If5ee905a15821296c355bb84cda38950ad8edc45
(cherry picked from commit a335f4c9adb0a00211345634f61917bdf5b412c2)

10 months agoMerge "CI: Allow runUnittests to use 'expect' to run the tests"
George Joseph [Sat, 17 Nov 2018 17:30:30 +0000 (11:30 -0600)]
Merge "CI: Allow runUnittests to use 'expect' to run the tests"

10 months agores/res_ari: Fix null endpoint handle
Sungtae Kim [Sat, 17 Nov 2018 02:33:20 +0000 (03:33 +0100)]
res/res_ari: Fix null endpoint handle

The res_ari(POST /channels/create handler) deos not check the endpoint
parameter length. And it causes core
dump.
Fixed it to check the parameter length. Also fixed memory leak.

ASTERISK-28169

Change-Id: Ibf10a9eb8a2e3a9ee1e13fbe748b2ecf955c3993

10 months agoMerge "core: Fix handling of restart from remote console."
George Joseph [Fri, 16 Nov 2018 15:22:26 +0000 (09:22 -0600)]
Merge "core: Fix handling of restart from remote console."

10 months agoCI: Allow runUnittests to use 'expect' to run the tests
George Joseph [Thu, 15 Nov 2018 17:41:44 +0000 (10:41 -0700)]
CI: Allow runUnittests to use 'expect' to run the tests

There seems to be a race condition between starting the asterisk
daemon and attempting to use 'asterisk -r' that can cause the
control socket file to not be created.  Since all of the Jenkins
slaves have 'expect' installed, the runUnittests script can use
it to start asterisk in the forground and issue the commands
interactively.  This is much more reliable and it can also make
startup errors more visible since they'll be in the Jenkins console
output.

If 'expect' isn't installed, the original daemon/asterisk -r
process is used.

Also added a "core show settings" before running the tests
and added "notice,warning,error" to the console log.

Change-Id: Idd656085f854afede813ac241b9e312b31358160

10 months agotaskprocessor: Prevent race creating new taskprocessor.
Corey Farrell [Mon, 12 Nov 2018 18:23:34 +0000 (13:23 -0500)]
taskprocessor: Prevent race creating new taskprocessor.

Task processors are retrieved using a 'get or create' pattern.  The
singleton container was unlocked between the get and create steps so
it's possible that two threads could create task processors with the
same name at the same time.

Change-Id: Id64fae94a6a1e940ddf38fde622dcd4391635382

10 months agopjproject-bundled: Use AST_DEVMODE for conditional compilation.
Corey Farrell [Fri, 16 Nov 2018 12:20:11 +0000 (07:20 -0500)]
pjproject-bundled: Use AST_DEVMODE for conditional compilation.

We previously allowed resample and g711 codecs to be built when
TEST_FRAMEWORK was enabled.  This could cause errors if the testsuite
was run without this option enabled.  Switch the build system to allow
those codecs to be built when --enable-dev-mode is used.  This removes a
chance for strange testsuite errors from use of an inadequate pjsua
binary.

Change-Id: Iee8a3613cdb711fa7e7d217c5a775a575907ae22

10 months agores_pjsip_caller_id: Use static pj_str_t for fromto header names.
Corey Farrell [Thu, 15 Nov 2018 20:47:50 +0000 (15:47 -0500)]
res_pjsip_caller_id: Use static pj_str_t for fromto header names.

PJSIP assumes that these header names are not allocated, does not clone
the name strings when reusing headers.

Block unload of res_pjsip_caller_id until shutdown to ensure static
memory stays valid.  It was previously unsafe to unload while any
sessions are active.

Change-Id: I190854dea943d6e441cf03733f8a0da661aea27f

10 months agoMerge "pbx_config: Only the first [globals] section is seen."
George Joseph [Thu, 15 Nov 2018 13:48:20 +0000 (07:48 -0600)]
Merge "pbx_config: Only the first [globals] section is seen."

10 months agores/res_pjsip_nat: Fix logic for REINVITES
Torrey Searle [Wed, 24 Oct 2018 12:38:37 +0000 (14:38 +0200)]
res/res_pjsip_nat: Fix logic for REINVITES

The presence of Record-Route in re-invites is optional, thus it is
important to make sure the dialog doesn't have a routset before
rewriting the contact header.

ASTERISK-28129 #close

Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc

10 months agocore: Fix handling of restart from remote console.
Corey Farrell [Thu, 15 Nov 2018 11:33:11 +0000 (06:33 -0500)]
core: Fix handling of restart from remote console.

We cannot use need_el_end and SIGURG when restarting.  Instead we need
to run el_end within the SIGHUP restartnow handler.

ASTERISK-28158

Change-Id: Ia852276363c81bdcf1aa29eb4558c5c2fa1218a0

10 months agoAST-2018-010: Fix length of buffer needed for SRV and NAPTR results
George Joseph [Thu, 25 Oct 2018 15:25:58 +0000 (09:25 -0600)]
AST-2018-010: Fix length of buffer needed for SRV and NAPTR results

When dn_expand was being called on SRV and NAPTR results, the
return value was being used to calculate the size of the buffer
needed to store the host names.  Since dn_expand returns the
length of the COMPRESSED name the buffer could be too short
to hold the EXPANDED name.  The expanded name is NULL terminated
so using strlen() is the correct way to determine the length
actually needed for the buffer.

ASTERISK-28127
Reported by: Jan Hoffmann

patches:
  patch.diff submitted by janhoffmann (license 6986)

Change-Id: I4d35d6c431c6c6836cb61d37b1378cc47f0b414d

10 months agoMerge "core: Ensure that el_end is always run when needed."
Joshua Colp [Wed, 14 Nov 2018 13:06:38 +0000 (07:06 -0600)]
Merge "core: Ensure that el_end is always run when needed."

10 months agoMerge "taskprocessor: Do not use separate allocation for stats or name."
Joshua Colp [Wed, 14 Nov 2018 13:05:10 +0000 (07:05 -0600)]
Merge "taskprocessor: Do not use separate allocation for stats or name."

10 months agoMerge "jansson-bundled: Patch for off-nominal crash."
George Joseph [Tue, 13 Nov 2018 20:39:37 +0000 (14:39 -0600)]
Merge "jansson-bundled: Patch for off-nominal crash."

10 months agotest_res_pjsip_scheduler: Fix possible write after free in scheduler_policy.
Corey Farrell [Tue, 13 Nov 2018 16:51:00 +0000 (11:51 -0500)]
test_res_pjsip_scheduler: Fix possible write after free in scheduler_policy.

It's possible for a 4th task to be spawned before we cancel.  This
results in a write to the already freed test_data1.  Wait long enough to
verify success of the cancelation before freeing test_data1.

Change-Id: I057e2fcbe97f8a175e50890be89c28c20490a20f

10 months agobridge_native_rtp.c: Fail native bridge if no framing match.
Robert Cripps [Wed, 17 Oct 2018 13:48:13 +0000 (15:48 +0200)]
bridge_native_rtp.c: Fail native bridge if no framing match.

ASTERISK-28110 #close

Change-Id: Ic64b8fc6a140a93fbdb2f97550a40d0ff334e607

10 months agotaskprocessor: Do not use separate allocation for stats or name.
Corey Farrell [Mon, 12 Nov 2018 00:32:11 +0000 (19:32 -0500)]
taskprocessor: Do not use separate allocation for stats or name.

Merge storage for the stats object and name string into the main
allocation for struct ast_taskprocessor.

Change-Id: I74fe9a7f357f0e6d63152f163cf5eef6428218e1

10 months agoMerge "res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue"
Joshua Colp [Mon, 12 Nov 2018 11:38:44 +0000 (05:38 -0600)]
Merge "res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue"

10 months agocore: Ensure that el_end is always run when needed.
Corey Farrell [Sun, 11 Nov 2018 13:34:53 +0000 (08:34 -0500)]
core: Ensure that el_end is always run when needed.

* Ignore console=yes configuration option in remote console processes.
* Use new flag to tell consolethread to run el_end and exit when needed.

ASTERISK-28158

Change-Id: I9e23b31d4211417ddc88c6bbfd83ea4c9f3e5438

10 months agojansson-bundled: Patch for off-nominal crash.
Corey Farrell [Thu, 8 Nov 2018 21:37:34 +0000 (16:37 -0500)]
jansson-bundled: Patch for off-nominal crash.

pack_string crashed on non-NULL strings returned when s->has_error was
true if the string was the result of 's' format without '#', '%' or '+'.

Change-Id: Ic125df691d81ba2cbc413e37bdae657b304d20d0

10 months agopbx_config: Only the first [globals] section is seen.
Corey Farrell [Fri, 2 Nov 2018 11:38:19 +0000 (07:38 -0400)]
pbx_config: Only the first [globals] section is seen.

If multiple [globals] sections are used (for example via separate
included files), only the first one is processed.  This can result in
lost global variables when using a modular extensions.conf.

ASTERISK-28146 #close

Change-Id: Iaac810c0a7c4d9b1bf8989fcc041cdb910ef08a0

10 months agores_pjsip: Send a 503 response when overload state if reliable transport.
Chris-Savinovich [Tue, 6 Nov 2018 22:44:34 +0000 (16:44 -0600)]
res_pjsip: Send a 503 response when overload state if reliable transport.

When Asterisk's taskprocessors get overloaded we need to reduce the work
load. res_pjsip currently ignores new SIP requests and relies on SIP
retransmissions in the hope that the overload condition will clear soon
enough to handle the retransmitted SIP request.
This change adds the following code after ast_taskprocessor_alert_get()
has returned TRUE:
1- identifies transport type. If non-udp then send a 503 response
2- if transport type is udp/udp6 then ignore, as before.

Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836

10 months agoMerge "stasis: Clarify lifetime of topics."
Joshua Colp [Wed, 7 Nov 2018 12:33:54 +0000 (06:33 -0600)]
Merge "stasis: Clarify lifetime of topics."

10 months agores_pjsip: formatting error in documentation
Kevin Harwell [Tue, 6 Nov 2018 22:35:30 +0000 (16:35 -0600)]
res_pjsip: formatting error in documentation

The use of a '|' in the "global/debug" synopsis documentation caused the
generated html table on the wiki to add an extra column that included the
text after the pipe.

This patch replaces the pipe with a comma.

ASTERISK-28150

Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719

10 months agores_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue
Alexei Gradinari [Mon, 5 Nov 2018 18:44:28 +0000 (13:44 -0500)]
res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue

The current round-robin method does not take the current taskprocessor
load into consideration when distributing requests.  Using the least-size
method the request goes to the taskprocessor that is servicing the least
number of active tasks at the current time.

Longer running tasks with the round-robin method can delay processing
tasks.

* Change the algorithm from round-robin to least-size for picking the
PJSIP taskprocessor from the default serializer pool.

Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd

10 months agoMerge "chan_sip: Attempt ast_do_pickup in handle_invite_replaces"
George Joseph [Mon, 5 Nov 2018 15:32:55 +0000 (09:32 -0600)]
Merge "chan_sip:  Attempt ast_do_pickup in handle_invite_replaces"

10 months agostasis: Clarify lifetime of topics.
Joshua Colp [Mon, 5 Nov 2018 14:30:54 +0000 (14:30 +0000)]
stasis: Clarify lifetime of topics.

As mentioned in the comment I've added in the code there is no
ability to unsubscribe all subscribers from a topic and explicitly
destroy it. This is not currently a problem as we have two types of
topics:

Long lived topics which exist for the lifetime of the system.
Ephemeral topics which feed a long lived topic.

In the case of the ephemeral topics there is no subscriber which does
not have its lifetime managed by the same entity that has created
the topic. This ensures that when the topic is being unreferenced the
subscribers are also unsubscribed and destroyed, allowing the topic
to ultimately be destroyed as well.

Change-Id: Ic5e244da7b16b1895ba1fc5ece481ebba5809c9a

10 months agochan_sip: Attempt ast_do_pickup in handle_invite_replaces
Jasper Hafkenscheid [Tue, 9 Oct 2018 12:44:57 +0000 (14:44 +0200)]
chan_sip:  Attempt ast_do_pickup in handle_invite_replaces

When a call pickup is performed using and invite with replaces header
the ast_do_pickup method is attempted and a PICKUP stasis message is sent.

ASTERISK-28081 #close
Reported-by: Luit van Drongelen

Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a

10 months agocontrib/sip_to_pjsip: add a --quiet option to avoid prints
Pascal Cadotte Michaud [Fri, 26 Oct 2018 15:53:40 +0000 (11:53 -0400)]
contrib/sip_to_pjsip: add a --quiet option to avoid prints

Using the --quiet or -q option in conjonction with /dev/stdout as the output
file allow the output to be used as a valid configuration.

Given a script that generates a valid sip.conf I can pipe the output of that
script into `sip_to_pjsip.py -q /dev/stdin /dev/stdout`. This allow me to use
that piped command in my pjsip.conf using the `exec` command.

ASTERISK-28136

Change-Id: I7b0e2e90e2549f3f8e01dc96701f111b5874c88d

10 months agoMerge "res_pjsip: Add XML documentation for "use_callerid_contact""
George Joseph [Wed, 31 Oct 2018 18:58:52 +0000 (13:58 -0500)]
Merge "res_pjsip: Add XML documentation for "use_callerid_contact""

10 months agoMerge "alembic: Fix use_callerid_contact option add script."
George Joseph [Wed, 31 Oct 2018 18:58:31 +0000 (13:58 -0500)]
Merge "alembic: Fix use_callerid_contact option add script."

10 months agoMerge "pjsip: new endpoint's options to control Connected Line updates"
George Joseph [Wed, 31 Oct 2018 18:57:15 +0000 (13:57 -0500)]
Merge "pjsip: new endpoint's options to control Connected Line updates"

10 months agoMerge "contrib/sip_to_pjsip: handle setvar in conversion"
George Joseph [Wed, 31 Oct 2018 18:55:48 +0000 (13:55 -0500)]
Merge "contrib/sip_to_pjsip: handle setvar in conversion"

10 months agoMerge "chan_sip deprecation."
George Joseph [Wed, 31 Oct 2018 18:53:07 +0000 (13:53 -0500)]
Merge "chan_sip deprecation."

10 months agores_pjsip: Add XML documentation for "use_callerid_contact"
Joshua Colp [Wed, 31 Oct 2018 12:53:08 +0000 (12:53 +0000)]
res_pjsip: Add XML documentation for "use_callerid_contact"

ASTERISK-28087

Change-Id: I69d48813ec514f5ef06c6de994cba52630e0a3b4

10 months agoalembic: Fix use_callerid_contact option add script.
Richard Mudgett [Tue, 30 Oct 2018 15:52:28 +0000 (10:52 -0500)]
alembic: Fix use_callerid_contact option add script.

ASTERISK-28087

Change-Id: I046d018015427d0916fab571b5a4f5367476f729

10 months agopjsip: new endpoint's options to control Connected Line updates
Alexei Gradinari [Mon, 22 Oct 2018 16:49:37 +0000 (12:49 -0400)]
pjsip: new endpoint's options to control Connected Line updates

This patch adds new options 'trust_connected_line' and 'send_connected_line'
to the endpoint.

The option 'trust_connected_line' is to control if connected line updates
are accepted from this endpoint.

The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.

The default value is 'yes' for both options.

Change-Id: I16af967815efd904597ec2f033337e4333d097cd

10 months agocontrib/sip_to_pjsip: handle setvar in conversion
Pascal Cadotte Michaud [Sat, 27 Oct 2018 14:59:15 +0000 (10:59 -0400)]
contrib/sip_to_pjsip: handle setvar in conversion

Given a sip.conf with the following content:

setvar FOO=1
setvar BAR=42

I want my generated pjsip.conf to containt the following set_vars

set_var FOO=1
set_var BAR=42

in the matching endpoint section.

Change-Id: I6c822401fda4133c3b44bf31e655b4eb939d4d26

10 months agoMerge "res_pjsip_notify: improve realtime performance on CLI completion on the endpoint"
George Joseph [Mon, 29 Oct 2018 18:23:05 +0000 (13:23 -0500)]
Merge "res_pjsip_notify: improve realtime performance on CLI completion on the endpoint"

10 months agores_pjsip_notify: improve realtime performance on CLI completion on the endpoint
Alexei Gradinari [Fri, 26 Oct 2018 21:18:38 +0000 (17:18 -0400)]
res_pjsip_notify: improve realtime performance on CLI completion on the endpoint

The module 'res_pjsip_notify' inefficiently makes a lot of DB requests
on CLI completion on the endpoint.

For example if there are 10k endpoints the module makes 10k requests
of these 10k records.

Even if a part of the endpoint entered
the module makes the same 10k requests and then filtered them by itself.

This patch gathers endpoints container by prefix
and adds all gathered endpoints to completion at once.

ASTERISK-28137 #close

Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b

10 months agores_pjsip_session: add new flag use_callerid_contact
Torrey Searle [Tue, 2 Oct 2018 12:31:43 +0000 (14:31 +0200)]
res_pjsip_session: add new flag use_callerid_contact

Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header.  This allows chan_pjsip to have
the same behavour as chan_sip

ASTERISK-28087 #close

Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95

10 months agochan_sip deprecation.
Corey Farrell [Wed, 10 Oct 2018 12:09:15 +0000 (08:09 -0400)]
chan_sip deprecation.

This officially deprecates chan_sip in Asterisk 17+.  A warning is
printed upon startup or module load to tell users that they should
consider migrating.  chan_sip is still built by default but the default
modules.conf skips loading it at startup.

Very important to note we are not scheduling a time where chan_sip will
be removed.  The goal of this change is to accurately inform end users
of the current state of chan_sip and encourage movement to the fully
supported chan_pjsip.

Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93

10 months agoUPDATE.txt: Fix formatting to match previous files.
Corey Farrell [Thu, 25 Oct 2018 12:54:19 +0000 (08:54 -0400)]
UPDATE.txt: Fix formatting to match previous files.

Add 'Section:' headings and use '-' for bullet points.

Change-Id: I7e2be35601ac8fea53b90d926da564512b6716e4

10 months agoMerge "res_parking: Stop setting the deprecated PARKINGSLOT channel variable."
Joshua Colp [Thu, 25 Oct 2018 12:50:52 +0000 (07:50 -0500)]
Merge "res_parking: Stop setting the deprecated PARKINGSLOT channel variable."

10 months agoMerge "app_dial/queue/followme: 'I' options to block initial updates in both directions"
Joshua Colp [Thu, 25 Oct 2018 12:46:38 +0000 (07:46 -0500)]
Merge "app_dial/queue/followme: 'I' options to block initial updates in both directions"

10 months agoMerge "bridge_softmix: Add SDP "label" attribute to streams"
Joshua Colp [Thu, 25 Oct 2018 12:45:23 +0000 (07:45 -0500)]
Merge "bridge_softmix:  Add SDP "label" attribute to streams"

10 months agoMerge "say: Remove legacy language deprecation logic"
George Joseph [Thu, 25 Oct 2018 12:38:09 +0000 (07:38 -0500)]
Merge "say: Remove legacy language deprecation logic"

10 months agoMerge "logger.c: Fix default console logging when no logger.conf available."
George Joseph [Thu, 25 Oct 2018 12:37:49 +0000 (07:37 -0500)]
Merge "logger.c: Fix default console logging when no logger.conf available."

10 months agoMerge "modules.conf.sample: Update preload usage documentation."
Joshua Colp [Thu, 25 Oct 2018 11:56:29 +0000 (06:56 -0500)]
Merge "modules.conf.sample: Update preload usage documentation."

10 months agores_parking: Stop setting the deprecated PARKINGSLOT channel variable.
Sean Bright [Thu, 18 Oct 2018 19:51:29 +0000 (15:51 -0400)]
res_parking: Stop setting the deprecated PARKINGSLOT channel variable.

Change-Id: Ia155ce2a53d61556aa4685524d1b48cfacfa3a8b

10 months agoMerge "func_callerid: Remove deprecated CALLERPRES() function."
Joshua Colp [Thu, 25 Oct 2018 10:51:18 +0000 (05:51 -0500)]
Merge "func_callerid: Remove deprecated CALLERPRES() function."

10 months agoMerge "res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability"
Joshua Colp [Thu, 25 Oct 2018 10:51:02 +0000 (05:51 -0500)]
Merge "res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability"

10 months agologger.c: Fix default console logging when no logger.conf available.
Richard Mudgett [Thu, 18 Oct 2018 00:34:37 +0000 (19:34 -0500)]
logger.c: Fix default console logging when no logger.conf available.

Default logging was not setup correctly when there was no logger.conf.
This resulted in many expected log messages not actually getting out to
the console.

Change-Id: I542e61c03b2f630ff5327f9de5641d776c6fa70c

10 months agoapp_dial/queue/followme: 'I' options to block initial updates in both directions
Alexei Gradinari [Wed, 26 Sep 2018 20:05:59 +0000 (16:05 -0400)]
app_dial/queue/followme: 'I' options to block initial updates in both directions

The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates
from the called parties to the caller.

This patch also blocks updates in the other direction before call is
answered.

ASTERISK-27980

Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01

10 months agomodules.conf.sample: Update preload usage documentation.
Richard Mudgett [Mon, 22 Oct 2018 19:31:27 +0000 (14:31 -0500)]
modules.conf.sample: Update preload usage documentation.

Change-Id: Id449d4435c38148b56ac4cfd61ae4d90ac66bb90

10 months agobridge_softmix: Add SDP "label" attribute to streams
George Joseph [Tue, 16 Oct 2018 12:02:19 +0000 (06:02 -0600)]
bridge_softmix:  Add SDP "label" attribute to streams

Adding the "label" attribute used for participant info correlation
was previously done in app_confbridge but it wasn't working
correctly because it didn't have knowledge about which video
streams belonged to which channel.  Only bridge_softmix has that
data so now it's set when the bridge topology is changed.

ASTERISK-28107

Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499

10 months agoMerge "astobj2: Eliminate legacy container allocation macros."
George Joseph [Wed, 24 Oct 2018 13:30:08 +0000 (08:30 -0500)]
Merge "astobj2: Eliminate legacy container allocation macros."

10 months agofunc_callerid: Remove deprecated CALLERPRES() function.
Sean Bright [Thu, 18 Oct 2018 19:24:19 +0000 (15:24 -0400)]
func_callerid: Remove deprecated CALLERPRES() function.

Change-Id: Ia1b2b386505b3102136dab02c45eaaf09f0f89c5

10 months agores_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
Nick French [Wed, 18 Jul 2018 12:45:26 +0000 (07:45 -0500)]
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability

This change implements a few different generic things which were brought
on by Google Voice SIP.

1.  The concept of flow transports have been introduced.  These are
configurable transports in pjsip.conf which can be used to reference a
flow of signaling to a target.  These have runtime configuration that can
be changed by the signaling itself (such as Service-Routes and
P-Preferred-Identity).  When used these guarantee an individual connection
(in the case of TCP or TLS) even if multiple flow transports exist to the
same target.

2.  Service-Routes (RFC 3608) support has been added to the outbound
registration module which when received will be stored on the flow
transport and used for requests referencing it.

3.  P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been
added to the outbound registration module.  If a P-Associated-URI header
is received it will be used on requests as the P-Preferred-Identity.

4.  Configurable outbound extension support has been added to the outbound
registration module.  When set the extension will be placed in the
Supported header.

5.  Header parameters can now be configured on an outbound registration
which will be placed in the Contact header.

6.  Google specific OAuth / Bearer token authentication
(draft-ietf-sipcore-sip-authn-02) has been added to the outbound
registration module.

All functionality changes are controlled by pjsip.conf configuration
options and do not affect non-configured pjsip endpoints otherwise.

ASTERISK-27971 #close

Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58

10 months agoMerge "res_xmpp: Remove deprecated JabberStatus application."
George Joseph [Wed, 24 Oct 2018 12:47:58 +0000 (07:47 -0500)]
Merge "res_xmpp: Remove deprecated JabberStatus application."

10 months agoMerge "app_dial/app_queue: Update application option documentation"
George Joseph [Wed, 24 Oct 2018 12:47:19 +0000 (07:47 -0500)]
Merge "app_dial/app_queue: Update application option documentation"

10 months agoMerge "lock: Replace __ast_mutex_logger with private log_mutex_error."
George Joseph [Wed, 24 Oct 2018 12:46:11 +0000 (07:46 -0500)]
Merge "lock: Replace __ast_mutex_logger with private log_mutex_error."

10 months agosay: Remove legacy language deprecation logic
Sean Bright [Tue, 23 Oct 2018 12:37:44 +0000 (08:37 -0400)]
say: Remove legacy language deprecation logic

These language codes (tw, ge, mx, and cz) were deprecated in 1.6.2.

Change-Id: I18e4d2af2e83556fa91e39a7338030583ef05d50