asterisk/asterisk.git
4 years agocore: Improve/simplify handling of required headers.
Corey Farrell [Fri, 31 Mar 2017 17:09:38 +0000 (13:09 -0400)]
core: Improve/simplify handling of required headers.

* Report failures if configure finds a required header is missing.
* Deduplicate includes between asterisk.h, astmm.h and compat.h.
* Unconditionally include headers in compat.h if required elsewhere.

Change-Id: Ie67d0185ca71fbfb81c9bdfaebe46a49e3c56dc5

4 years agoMerge "sdp: Add support for setting connection address and clean up state."
Mark Michelson [Mon, 3 Apr 2017 14:32:08 +0000 (09:32 -0500)]
Merge "sdp: Add support for setting connection address and clean up state."

4 years agoMerge "Forward declare 'struct ast_json' in asterisk.h"
zuul [Sat, 1 Apr 2017 13:46:14 +0000 (08:46 -0500)]
Merge "Forward declare 'struct ast_json' in asterisk.h"

4 years agochan_vpb.cc: Fix compiler error.
Richard Mudgett [Fri, 31 Mar 2017 18:14:05 +0000 (13:14 -0500)]
chan_vpb.cc: Fix compiler error.

Added missing channel technology read/write stream callback
initialization.

Change-Id: I829043a327d987e0d964485dd3d27964bebbd623

4 years agoMerge "cdr_pgsql: Fix buffer overflow calling libpq"
zuul [Fri, 31 Mar 2017 14:01:23 +0000 (09:01 -0500)]
Merge "cdr_pgsql: Fix buffer overflow calling libpq"

4 years agoMerge "build: Fix deb build issues with fakeroot"
George Joseph [Fri, 31 Mar 2017 13:20:29 +0000 (08:20 -0500)]
Merge "build: Fix deb build issues with fakeroot"

4 years agoForward declare 'struct ast_json' in asterisk.h
Corey Farrell [Thu, 30 Mar 2017 23:28:18 +0000 (19:28 -0400)]
Forward declare 'struct ast_json' in asterisk.h

The ast_json structure is used in many Asterisk headers and is often the
only part of json.h used.  This adds a forward declaration to asterisk.h
and removes the include of json.h from many headers.  The declaration
has been left in endpoints.h and stasis.h to avoid problems with source
files that use ast_json functions without directly including json.h.

ari.h continues to include json.h as it uses enum
ast_json_encoding_format.

Change-Id: Id766aabce6bed56626d27e8d29f559b5e687b769

4 years agoMerge "CEL: Remove header declarations of non-existant functions."
zuul [Thu, 30 Mar 2017 23:29:09 +0000 (18:29 -0500)]
Merge "CEL: Remove header declarations of non-existant functions."

4 years agocdr_pgsql: Fix buffer overflow calling libpq
Sean Bright [Thu, 30 Mar 2017 13:11:46 +0000 (09:11 -0400)]
cdr_pgsql: Fix buffer overflow calling libpq

Implement the same buffer size checking done in cel_pgsql.

ASTERISK-26896 #close
Reported by: twisted

Change-Id: Iaacfa1f1de7cb1e9414d121850d2d8c2888f3f48

4 years agobuild: Fix deb build issues with fakeroot
Walter Doekes [Tue, 28 Mar 2017 18:01:16 +0000 (20:01 +0200)]
build: Fix deb build issues with fakeroot

If DESTDIR is set, don't call ldconfig. Assume that DESTDIR is used to
create a binary archive. The ldconfig call should be delegated to the
archive postinst script. This fixes the case where fakeroot wraps 'make
install' causing $EUID to be 0 even though it doesn't have permission to
call ldconfig.

The previous logic in configure.ac to detect and correct libdir
has been removed as it was not completely accurate.  CentOS 64-bit
users should again specifiy --libdir=/usr/lib64 when configuring
to prevent install to /usr/lib.

Updated Makefile:check-old-libdir to check for orphans in
lib64 when installing to lib as well as orphans in lib when installing
to lib64.

Updated Makefile and main/Makefile uninstall targets to remove the
orphans using the new logic.

ASTERISK-26705

Change-Id: I51739d4a03e60bff38be719b8d2ead0007afdd51

4 years agoMerge "res_pjsip_config_wizard: Add 2 new parameters to help with proxy config"
zuul [Thu, 30 Mar 2017 22:02:38 +0000 (17:02 -0500)]
Merge "res_pjsip_config_wizard: Add 2 new parameters to help with proxy config"

4 years agosdp: Add support for setting connection address and clean up state.
Joshua Colp [Mon, 27 Mar 2017 20:32:45 +0000 (20:32 +0000)]
sdp: Add support for setting connection address and clean up state.

This change cleans up state management for media streams by moving
RTP instances into their own session structure and adding additional
details that are not relevant to the core (such as connection address).
These can live either in the local capabilities or joint capabilities.

The ability to set explicit connection address information for
the purposes of direct media and NAT has also been added at the
global and stream specific level.

ASTERISK-26900

Change-Id: If7e5307239a9534420732de11c451a2705b6b681

4 years agoastobj2: Prevent potential deadlocks with ao2_global_obj_release
Sean Bright [Wed, 29 Mar 2017 15:11:51 +0000 (11:11 -0400)]
astobj2: Prevent potential deadlocks with ao2_global_obj_release

The ao2_global_obj_release() function holds an exclusive lock on the
global object while it is being dereferenced. Any destructors that
run during this time that call ao2_global_obj_ref() will deadlock
because a read lock is required.

Instead, we make the global object inaccessible inside of the write
lock and only dereference it once we have released the lock. This
allows the affected destructors to fail gracefully.

While this doesn't completely solve the referenced issue (the error
message about not being able to create an IQ continues to be shown)
it does solve the backtrace spew that accompanied it.

ASTERISK-21009 #close
Reported by: Marcello Ceschia

Change-Id: Idf40ae136b5070dba22cb576ea8414fbc9939385

4 years agoCEL: Remove header declarations of non-existant functions.
Corey Farrell [Thu, 30 Mar 2017 15:18:38 +0000 (11:18 -0400)]
CEL: Remove header declarations of non-existant functions.

ast_cel_alloc and ast_cel_destroy do not exist in code, remove them from
the headers.

Change-Id: I99ce848e2e109e7d61771559f559b9e57973e45c

4 years agoMerge "cel_pgsql.c: Fix buffer overflow calling libpq"
Joshua Colp [Thu, 30 Mar 2017 10:12:11 +0000 (05:12 -0500)]
Merge "cel_pgsql.c: Fix buffer overflow calling libpq"

4 years agoMerge "srtp: Allow zero as tag value for a sRTP Crypto Suite."
Joshua Colp [Wed, 29 Mar 2017 22:49:55 +0000 (17:49 -0500)]
Merge "srtp: Allow zero as tag value for a sRTP Crypto Suite."

4 years agoMerge "Add DTLS sanity check."
zuul [Wed, 29 Mar 2017 21:11:12 +0000 (16:11 -0500)]
Merge "Add DTLS sanity check."

4 years agoMerge "core: Remove embedded module support"
George Joseph [Wed, 29 Mar 2017 19:40:36 +0000 (14:40 -0500)]
Merge "core: Remove embedded module support"

4 years agoMerge "channel: Remove old epoll support and fixed max number of file descriptors."
zuul [Wed, 29 Mar 2017 17:45:47 +0000 (12:45 -0500)]
Merge "channel: Remove old epoll support and fixed max number of file descriptors."

4 years agoMerge "alembic: Turn off execute bit on non-executable python scripts"
George Joseph [Wed, 29 Mar 2017 16:32:38 +0000 (11:32 -0500)]
Merge "alembic: Turn off execute bit on non-executable python scripts"

4 years agoMerge "res_musiconhold: Don't chdir() when scanning MoH files"
zuul [Wed, 29 Mar 2017 15:11:01 +0000 (10:11 -0500)]
Merge "res_musiconhold: Don't chdir() when scanning MoH files"

4 years agocel_pgsql.c: Fix buffer overflow calling libpq
Josh Roberson [Mon, 27 Mar 2017 16:49:08 +0000 (11:49 -0500)]
cel_pgsql.c: Fix buffer overflow calling libpq

PQEscapeStringConn() expects the buffer passed in to be an
adequitely sized buffer to write out the escaped SQL value string
into.  It is possible, for large values (such as large values to
Dial with a lot of devices) to have more than our 512+1 byte
allocation and thus cause libpq to create a buffer overrun.

glibc will nicely ABRT asterisk for you, citing a stack smash.

Let's only allocate it to be as large as needed:
If we have a value, then (strlen(value) * 2) + 1 (as recommended
by libpq), and if we have none, just one byte to hold our null
will do.

ASTERISK-26896 #close

Change-Id: If611c734292618ed68dde17816d09dd16667dea2

4 years agosrtp: Allow zero as tag value for a sRTP Crypto Suite.
Alexander Traud [Wed, 29 Mar 2017 13:04:05 +0000 (15:04 +0200)]
srtp: Allow zero as tag value for a sRTP Crypto Suite.

ASTERISK-25490 #close

Change-Id: I1c5fc0942c33c96d62b24203aad0f1e1a1a0131f

4 years agoMerge "res_xmpp: Fix ref counting issue"
Joshua Colp [Wed, 29 Mar 2017 11:57:48 +0000 (06:57 -0500)]
Merge "res_xmpp: Fix ref counting issue"

4 years agores_pjsip_config_wizard: Add 2 new parameters to help with proxy config
George Joseph [Tue, 28 Mar 2017 18:10:32 +0000 (12:10 -0600)]
res_pjsip_config_wizard: Add 2 new parameters to help with proxy config

Two new parameters have been added to the pjsip config wizard.

 * Setting 'sends_line_with_registrations' to true will cause the wizard
   to skip the creation of an identify object to match incoming request
   to the endpoint and instead add the line and endpoint parameters to
   the outbound registration object.

 * Setting 'outbound_proxy' is a shortcut for adding individual
   endpoint/outbound_proxy, aor/outbound_proxy and
   registration/outbound_proxy parameters.

Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0
(cherry picked from commit a827892ff77cd37912b528d9c45b446be091bbc0)
(cherry picked from commit 27344675be1941d30508c6e6bd684acdd0791e1a)

4 years agoMerge "res_xmpp: Use incremental backoff when a read error occurs"
Joshua Colp [Tue, 28 Mar 2017 21:46:52 +0000 (16:46 -0500)]
Merge "res_xmpp: Use incremental backoff when a read error occurs"

4 years agoalembic: Turn off execute bit on non-executable python scripts
Sean Bright [Tue, 28 Mar 2017 14:29:25 +0000 (10:29 -0400)]
alembic: Turn off execute bit on non-executable python scripts

Change-Id: I744c986da4a38aeff8c00837eb89de7841fbc86c

4 years agoAdd DTLS sanity check.
Richard Mudgett [Mon, 27 Mar 2017 17:37:39 +0000 (12:37 -0500)]
Add DTLS sanity check.

Change-Id: Ib32612cf6c7ce9213a11b9cba82f630f8cd3564b

4 years agochannel: Remove old epoll support and fixed max number of file descriptors.
Joshua Colp [Wed, 8 Mar 2017 13:24:46 +0000 (13:24 +0000)]
channel: Remove old epoll support and fixed max number of file descriptors.

This change removes the old epoll support which has not been used or
maintained in quite some time.

The fixed number of file descriptors on a channel has also been removed.
File descriptors are now contained in a growable vector. This can be
used like before by specifying a specific position to store a file
descriptor at or using a new API call, ast_channel_fd_add, which adds
a file descriptor to the channel and returns its position.

Tests have been added which cover the growing behavior of the vector
and the new API call.

ASTERISK-26885

Change-Id: I1a754b506c009b83dfdeeb08c2d2815db30ef928

4 years agores_musiconhold: Document the 'format' option
Sean Bright [Mon, 27 Mar 2017 14:35:15 +0000 (10:35 -0400)]
res_musiconhold: Document the 'format' option

ASTERISK-26086 #close
Reported by: Jens B├╝rger

Change-Id: I6aab666c0bf01fd0c64d7a5bcb22fa7f5d41335e

4 years agocore: Remove embedded module support
Sean Bright [Fri, 24 Mar 2017 12:43:05 +0000 (08:43 -0400)]
core: Remove embedded module support

This has not worked for some time and is no longer actively maintained.

Change-Id: I5110b0db69c152761b58fa025cb0a53b0e544d99

4 years agores_musiconhold: Don't chdir() when scanning MoH files
Sean Bright [Mon, 27 Mar 2017 13:58:17 +0000 (09:58 -0400)]
res_musiconhold: Don't chdir() when scanning MoH files

There doesn't appear to be any reason that we are chdir'ing in
moh_scan_files, and in the event of an Asterisk crash, the core files
may not get written because we have changed into a read-only directory.

ASTERISK-23996 #close
Reported by: Walter Doekes

Change-Id: Iac806dce01b3335963fbd62d4b4da9a65c614354

4 years agoMerge "res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts"
Joshua Colp [Sat, 25 Mar 2017 10:20:03 +0000 (05:20 -0500)]
Merge "res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts"

4 years agoMerge "cdr: Allow setting of user field from 'h' extension"
Joshua Colp [Fri, 24 Mar 2017 23:02:12 +0000 (18:02 -0500)]
Merge "cdr: Allow setting of user field from 'h' extension"

4 years agoMerge "rtp_engine: allocate RTP dynamic payloads per session"
zuul [Fri, 24 Mar 2017 21:22:55 +0000 (16:22 -0500)]
Merge "rtp_engine: allocate RTP dynamic payloads per session"

4 years agores_xmpp: Use incremental backoff when a read error occurs
Sean Bright [Thu, 23 Mar 2017 14:48:40 +0000 (10:48 -0400)]
res_xmpp: Use incremental backoff when a read error occurs

If a read error occurs, we immediately attempt a reconnect without any
delay. Instead, let's sleep and backoff up to 60 seconds before we try
again.

ASTERISK-24712 #close
Reported by: Matthias Urlichs

Change-Id: I6fe10ef4734837727437beab715e336777f13f48

4 years agoMerge "pjproject_bundled: raise timeout value used when downloading"
zuul [Fri, 24 Mar 2017 20:42:48 +0000 (15:42 -0500)]
Merge "pjproject_bundled: raise timeout value used when downloading"

4 years agoMerge "res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus"
zuul [Fri, 24 Mar 2017 17:25:07 +0000 (12:25 -0500)]
Merge "res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus"

4 years agoMerge "res_xmpp: Include client name in connection related error messages"
zuul [Fri, 24 Mar 2017 16:55:38 +0000 (11:55 -0500)]
Merge "res_xmpp: Include client name in connection related error messages"

4 years agores_pjsip_sdp_rtp: Set hangup cause for RTP timeouts
Sean Bright [Fri, 24 Mar 2017 16:29:10 +0000 (12:29 -0400)]
res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts

chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL
(44) when a channel is hung up due to an RTP timeout. So do the same
when it happens with PJSIP for parity.

Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8

4 years agoMerge "res_xmpp: Don't crash when trying to send a message without a connection"
Joshua Colp [Fri, 24 Mar 2017 15:46:34 +0000 (10:46 -0500)]
Merge "res_xmpp: Don't crash when trying to send a message without a connection"

4 years agoMerge "res_xmpp: Correctly check return value of SSL_connect"
zuul [Fri, 24 Mar 2017 14:13:06 +0000 (09:13 -0500)]
Merge "res_xmpp: Correctly check return value of SSL_connect"

4 years agoMerge "res_xmpp: Try to provide useful errors messages from OpenSSL"
zuul [Fri, 24 Mar 2017 14:12:57 +0000 (09:12 -0500)]
Merge "res_xmpp: Try to provide useful errors messages from OpenSSL"

4 years agoMerge "audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor."
Joshua Colp [Fri, 24 Mar 2017 12:25:07 +0000 (07:25 -0500)]
Merge "audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor."

4 years agoAMI: Updated version
Kevin Harwell [Thu, 23 Mar 2017 19:01:40 +0000 (14:01 -0500)]
AMI: Updated version

Updated the AMI version for the following reason (see CHANGES for more details):

The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now
contains a new optional parameter, 'MatchHeader'.

Change-Id: Ie206913ef1dcfa6a2ebe3282da2387e52d6f05b9

4 years agopjproject_bundled: raise timeout value used when downloading
Kevin Harwell [Thu, 23 Mar 2017 17:07:09 +0000 (12:07 -0500)]
pjproject_bundled: raise timeout value used when downloading

After configuring Asterisk with '--with-pjproject-bundled' the configure/build
process attempts to download pjproject from its download site. Currently, a
timeout of 10 seconds is used that will stop the download process if pjproject
has not been fully downloaded in that time. For some systems this was not enough
time and the process was timing out too early.

This patch raises the download timeout value to '60'. Also, this patch fixes
another bug where the DOWNLOAD_TIMEOUT variable was not being properly exported
due to a naming error. DOWNLOAD_MAX_TIMEOUT is now properly renamed to
DOWNLOAD_TIMEOUT.

ASTERISK-26814 #close

Change-Id: Ia56e4e8a3d39db76bc8a1852b2cf07ec10b39842

4 years agores_xmpp: Correct implementation of JABBER_STATUS & JabberStatus
Sean Bright [Thu, 23 Mar 2017 01:33:02 +0000 (21:33 -0400)]
res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus

The documentation for JABBER_STATUS (and the deprecated JabberStatus
app) indicate that a return value of 7 indicates that the specified
buddy was not in the roster. It also indicates that you can specify a
"bare" JID (one without a resource). Unfortunately the actual behavior
does not match the documented behavior.

Assuming that our roster includes the buddy online and available
"valid@example.org/Valid" and does *not* include the buddy
"invalid@example.org", the JABBER_STATUS() function returns the
following before this patch:

+------------------------------+------------+--------------------------+
| Buddy                        | Status     | Result                   |
+------------------------------+------------+--------------------------+
| valid@example.org            |  Online    |  7 (Not in roster)       |
| valid@example.org/Valid      |  Online    |  1 (Online)              |
| valid@example.org/Invalid    |  N/A       |  7 (Not in roster)       |
| invalid@example.org          |  N/A       |  Error logged, no return |
| invalid@example.org/Valid    |  N/A       |  Error logged, no return |
+------------------------------+------------+--------------------------+

And after this patch:

+------------------------------+------------+--------------------------+
| Buddy                        | Status     | Result                   |
+------------------------------+------------+--------------------------+
| valid@example.org            |  Online    |  1 (Online)              |
| valid@example.org/Valid      |  Online    |  1 (Online)              |
| valid@example.org/Invalid    |  N/A       |  6 (Offline)             |
| invalid@example.org          |  N/A       |  7 (Not in roster)       |
| invalid@example.org/Valid    |  N/A       |  7 (Not in roster)       |
+------------------------------+------------+--------------------------+

This brings the behavior in line with the documentation.

ASTERISK-23510 #close
Reported by: Anthony Critelli

Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf

4 years agores_xmpp: Try to provide useful errors messages from OpenSSL
Sean Bright [Thu, 23 Mar 2017 14:45:35 +0000 (10:45 -0400)]
res_xmpp: Try to provide useful errors messages from OpenSSL

If any errors occur during the TLS connection setup, we currently dump a
fairly generic error message. So instead we try to pull in something
useful from OpenSSL to report instead.

ASTERISK-24712
Reported by: Matthias Urlichs

Change-Id: I288500991a9681f447d92913b11fedaf426087f4

4 years agores_xmpp: Fix ref counting issue
Sean Bright [Thu, 23 Mar 2017 10:19:18 +0000 (06:19 -0400)]
res_xmpp: Fix ref counting issue

The only remaining reference to the endpoint is in the endpoints
container, and because it is unlinked in ast_endpoint_shutdown, we don't
have to explicitly cleanup the endpoint ourselves.

Change-Id: I912a2692e52d3e2ed445b32d8ae3f9004bc2f2e8

4 years agores_xmpp: Correctly check return value of SSL_connect
Sean Bright [Thu, 23 Mar 2017 14:30:18 +0000 (10:30 -0400)]
res_xmpp: Correctly check return value of SSL_connect

SSL_connect returns non-zero for both success and some error conditions
so simply negating is inadequate.

Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1

4 years agores_xmpp: Don't crash when trying to send a message without a connection
Sean Bright [Wed, 22 Mar 2017 22:32:37 +0000 (18:32 -0400)]
res_xmpp: Don't crash when trying to send a message without a connection

If we never establish a connection to our Jabber server, iksemel never sets up
its internal transport pointer, so attempting to send a message dereferences a
NULL pointer and causes a crash.

ASTERISK-21855 #close
Reported by: Jeremy Kister

Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c

4 years agores_xmpp: Include client name in connection related error messages
Sean Bright [Wed, 22 Mar 2017 20:40:29 +0000 (16:40 -0400)]
res_xmpp: Include client name in connection related error messages

ASTERISK-25622 #close
Reported by: Sean Darcy

Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9

4 years agoMerge "res_pjsip_session: Enable RFC3578 overlap dialing support."
Joshua Colp [Wed, 22 Mar 2017 22:08:08 +0000 (17:08 -0500)]
Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."

4 years agoMerge "CHANNEL(callid): Give dialplan access to the callid."
Joshua Colp [Wed, 22 Mar 2017 20:49:42 +0000 (15:49 -0500)]
Merge "CHANNEL(callid): Give dialplan access to the callid."

4 years agortp_engine: allocate RTP dynamic payloads per session
Kevin Harwell [Mon, 20 Mar 2017 18:27:31 +0000 (13:27 -0500)]
rtp_engine: allocate RTP dynamic payloads per session

Dynamic payload types were statically defined in Asterisk. This unfortunately
limited the number of dynamic payloads that could be registered. With this patch
dynamic payload type numbers are now assigned dynamically and per RTP instance.
However, in order to limit any issues where some clients expect the old
statically defined value this patch makes it so the value Asterisk used to pre-
designate is used for the dynamic assignment if available.

An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf)
that turns the new dynamic behavior on or off. When off it reverts back to using
statically defined payload values. This option defaults to "yes" in Asterisk 15.

ASTERISK-26515 #close
patches:
  ASTERISK-26515.diff submitted by jcolp (license 5000

Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc

4 years agoMerge "res_pjsip_messaging: Check URI type before dereferencing"
zuul [Wed, 22 Mar 2017 17:36:43 +0000 (12:36 -0500)]
Merge "res_pjsip_messaging: Check URI type before dereferencing"

4 years agoMerge "Revert "app_queue: Handle the caller being redirected out of a queue bridge""
zuul [Wed, 22 Mar 2017 15:54:56 +0000 (10:54 -0500)]
Merge "Revert "app_queue: Handle the caller being redirected out of a queue bridge""

4 years agoMerge "app_queue: Member stuck as pending after forwarding previous call from queue"
zuul [Wed, 22 Mar 2017 14:50:22 +0000 (09:50 -0500)]
Merge "app_queue: Member stuck as pending after forwarding previous call from queue"

4 years agocdr: Allow setting of user field from 'h' extension
Sebastian Gutierrez [Tue, 21 Mar 2017 17:32:06 +0000 (14:32 -0300)]
cdr: Allow setting of user field from 'h' extension

The CDR code previously did not allow the user field to be set
from the 'h' extension in the dialplan. This change removes that
limitation and allows it to be set.

ASTERISK-26818

Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6

4 years agoMerge "pjsip: prevent memory corruption on creation of xml bodies"
zuul [Wed, 22 Mar 2017 13:32:06 +0000 (08:32 -0500)]
Merge "pjsip: prevent memory corruption on creation of xml bodies"

4 years agores_pjsip_session: Enable RFC3578 overlap dialing support.
Richard Begg [Tue, 14 Mar 2017 21:45:06 +0000 (08:45 +1100)]
res_pjsip_session: Enable RFC3578 overlap dialing support.

Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched
destinations) as currently provided by chan_sip is missing from res_pjsip.
This patch adds a new endpoint attribute (allow_overlap) [defaults to yes]
which when set to yes enables 484 responses to partial destination
matches rather than the current 404.

ASTERISK-26864

Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6

4 years agoMerge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references."
zuul [Wed, 22 Mar 2017 02:51:49 +0000 (21:51 -0500)]
Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references."

4 years agoMerge "res_hep: Capture actual transport type in use"
zuul [Wed, 22 Mar 2017 00:47:16 +0000 (19:47 -0500)]
Merge "res_hep: Capture actual transport type in use"

4 years agores_hep: Capture actual transport type in use
Sean Bright [Tue, 21 Mar 2017 11:59:12 +0000 (07:59 -0400)]
res_hep: Capture actual transport type in use

Rather than hard-coding UDP, allow consumers of the HEP API to specify
which protocol is in use. Update the PJSIP provider to pass in the
current protocol type.

ASTERISK-26850 #close

Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978

4 years agoRevert "app_queue: Handle the caller being redirected out of a queue bridge"
Sean Bright [Tue, 21 Mar 2017 14:57:46 +0000 (08:57 -0600)]
Revert "app_queue: Handle the caller being redirected out of a queue bridge"

This reverts commit 163e9e53dc7d84dd42721e733b7706c8147bdd27.

Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b

4 years agores_pjsip_messaging: Check URI type before dereferencing
Sean Bright [Tue, 21 Mar 2017 13:26:28 +0000 (09:26 -0400)]
res_pjsip_messaging: Check URI type before dereferencing

We aren't validating that the URI we just parsed is a SIP/SIPS one before
trying to access the user, host, and port members of a possibly uninitialized
structure.

Also update the MessageSend documentation to indicate what 'from' formats are
accepted.

ASTERISK-26484 #close
Reported by: Vinod Dharashive

Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30

4 years agopjsip: prevent memory corruption on creation of xml bodies
Joshua Elson [Mon, 13 Mar 2017 20:21:23 +0000 (14:21 -0600)]
pjsip: prevent memory corruption on creation of xml bodies

ASTERISK-26776 #close

Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2

4 years agobridge_softmix: Ignore non-voice frames from translator
Sean Bright [Mon, 20 Mar 2017 21:27:24 +0000 (17:27 -0400)]
bridge_softmix: Ignore non-voice frames from translator

Some codecs - codec_speex specifically - take voice frames and return
other types of frames, like CNG. If we subsequently treat those as
voice frames, we'll run into trouble when destroying the frame because
of the requirement that each voice frame have an associated format.

ASTERISK-26880 #close
Reported by: Kirsty Tyerman

Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c

4 years agoMerge "res/res_pjsip_session: Only check localnet if it is defined"
Joshua Colp [Mon, 20 Mar 2017 19:39:20 +0000 (14:39 -0500)]
Merge "res/res_pjsip_session: Only check localnet if it is defined"

4 years agoaudiohook.c: Lost RTP packets lead to out-of-sync MixMonitor.
Aaron An [Wed, 15 Mar 2017 04:49:12 +0000 (12:49 +0800)]
audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor.

Fixed a bug in function "ast_audiohook_write_frame" that checked the
variable other_factory_samples and only flushed the factories, so they
would be in sync, when other_factory_samples > 0. When there is not any
rtp incoming the variable other_factory_samples will be 0, and although
the result of "our_factory_ms - other_factory_ms" may be very large,
this led to the record file not syncing.

ASTERISK-26875 #close
Reported-by: Aaron An
Tested-by: Aaron An

Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22

4 years agoMerge "thread safety: Don't use getprotobyname()"
zuul [Mon, 20 Mar 2017 18:07:51 +0000 (13:07 -0500)]
Merge "thread safety: Don't use getprotobyname()"

4 years agothread safety: Don't use getprotobyname()
Sean Bright [Sat, 18 Mar 2017 17:30:32 +0000 (13:30 -0400)]
thread safety: Don't use getprotobyname()

POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.

Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.

Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48

4 years agores_rtp_asterisk: Pass correct data length to ast_rtcp_interpret
Sean Bright [Sun, 19 Mar 2017 18:26:38 +0000 (14:26 -0400)]
res_rtp_asterisk: Pass correct data length to ast_rtcp_interpret

We are currently passing in the capacity of the read buffer instead of the
number of bytes that we actually read off the wire.

Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36

4 years agoMerge "app_queue: Fix locking behavior in stasis message handlers"
Joshua Colp [Sat, 18 Mar 2017 11:53:08 +0000 (06:53 -0500)]
Merge "app_queue: Fix locking behavior in stasis message handlers"

4 years agoMerge "chan_sip: Add rtcp-mux support"
Joshua Colp [Sat, 18 Mar 2017 10:38:19 +0000 (05:38 -0500)]
Merge "chan_sip: Add rtcp-mux support"

4 years agoMerge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped."
Joshua Colp [Sat, 18 Mar 2017 10:37:29 +0000 (05:37 -0500)]
Merge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped."

4 years agoMerge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed."
Joshua Colp [Sat, 18 Mar 2017 10:36:34 +0000 (05:36 -0500)]
Merge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed."

4 years agoMerge "app_confbridge: Fix ConfbridgeTalking AMI event description."
Joshua Colp [Sat, 18 Mar 2017 00:49:21 +0000 (19:49 -0500)]
Merge "app_confbridge: Fix ConfbridgeTalking AMI event description."

4 years agoMerge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error"
Joshua Colp [Fri, 17 Mar 2017 19:45:05 +0000 (14:45 -0500)]
Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error"

4 years agoMerge "res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport"
Joshua Colp [Fri, 17 Mar 2017 16:47:36 +0000 (11:47 -0500)]
Merge "res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport"

4 years agoapp_queue: Member stuck as pending after forwarding previous call from queue
Robert Mordec [Tue, 14 Mar 2017 14:27:56 +0000 (15:27 +0100)]
app_queue: Member stuck as pending after forwarding previous call from queue

Queue member will get stuck in pending_members if queue calls a device
that is different from the one observed for state changes.

This patch removes members from pending_members as a result of channel stasis
events such as blind or attended transfers and hangup.

ASTERISK-26862 #close

Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727

4 years agoCHANNEL(callid): Give dialplan access to the callid.
Richard Mudgett [Thu, 23 Feb 2017 05:26:13 +0000 (23:26 -0600)]
CHANNEL(callid): Give dialplan access to the callid.

* Added CHANNEL(callid) to retrieve the call identifier log tag associated
with the channel.  Dialplan now has access to the call log search key
associated with the channel so it can be saved in case there is a problem
with the call.

ASTERISK-26878

Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f

4 years agoapp_queue: Fix locking behavior in stasis message handlers
Sean Bright [Thu, 16 Mar 2017 13:42:54 +0000 (09:42 -0400)]
app_queue: Fix locking behavior in stasis message handlers

The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.

Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088

4 years agochan_sip: Add rtcp-mux support
Sean Bright [Wed, 8 Mar 2017 01:28:18 +0000 (20:28 -0500)]
chan_sip: Add rtcp-mux support

ASTERISK-26846 #close

Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639

4 years agoapp_confbridge: Fix ConfbridgeTalking AMI event description.
Richard Mudgett [Thu, 16 Mar 2017 21:50:17 +0000 (16:50 -0500)]
app_confbridge: Fix ConfbridgeTalking AMI event description.

Thanks to Chris Howard for pointing this out on the wiki.

Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705

4 years agores_pjsip_asterisk.c: Fix compile error if libsrtp is not installed.
Richard Mudgett [Thu, 16 Mar 2017 21:37:42 +0000 (16:37 -0500)]
res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed.

struct ast_rtcp does not define the dtls member if SRTP is not enabled.

ASTERISK-26732

Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e

4 years agoMerge "res_pjsip: Symmetric transports"
Joshua Colp [Thu, 16 Mar 2017 21:04:43 +0000 (16:04 -0500)]
Merge "res_pjsip:  Symmetric transports"

4 years agores_pjsip_sdp_rtp.c: Fix cut-n-paste error
Richard Mudgett [Thu, 16 Mar 2017 20:45:57 +0000 (15:45 -0500)]
res_pjsip_sdp_rtp.c: Fix cut-n-paste error

We were inadvertenly referencing the cos_video option to determine if we
should set the tos_audio and cos_audio value on the RTP instance.

Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0

4 years agores/res_pjsip_session: Only check localnet if it is defined
Matt Jordan [Thu, 16 Mar 2017 15:39:00 +0000 (10:39 -0500)]
res/res_pjsip_session: Only check localnet if it is defined

If local_net is not defined on a transport, transport_state->localnet
will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in
this case, causing the external_media_address, if set, to be skipped.

This patch causes us to only check if we are sending within a network if
local_net is defined.

ASTERISK-26879 #close

Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb

4 years agoMerge "RFC sdp: Initial SDP creation"
Joshua Colp [Thu, 16 Mar 2017 19:45:20 +0000 (14:45 -0500)]
Merge "RFC sdp: Initial SDP creation"

4 years agores_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
Richard Begg [Tue, 14 Mar 2017 21:22:42 +0000 (08:22 +1100)]
res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport

Currently a wildcard address is used for the local RTP socket, which
will not always result in the same address as used by the SIP socket
(e.g. if explicit transport addresses are configured).
Use the transport's host address when binding new local RTP sockets if
available.

ASTERISK-26851

Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a

4 years agores_pjsip: Symmetric transports
George Joseph [Tue, 7 Mar 2017 14:33:26 +0000 (07:33 -0700)]
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

4 years agores_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped.
Joshua Colp [Thu, 16 Mar 2017 14:07:55 +0000 (14:07 +0000)]
res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped.

This change removes an assumption that when DTLS is stopped
an RTCP session will be present on the RTP session. This is not
always the case.

ASTERISK-26732

Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611

4 years agoMerge "Add rtcp-mux support"
Joshua Colp [Thu, 16 Mar 2017 15:46:01 +0000 (10:46 -0500)]
Merge "Add rtcp-mux support"

4 years agoMerge "chan_iax2: Reload of iax peer results in loss of host address/port"
Joshua Colp [Thu, 16 Mar 2017 10:23:37 +0000 (05:23 -0500)]
Merge "chan_iax2: Reload of iax peer results in loss of host address/port"

4 years agoMerge "res/res_pjsip_refer: call xfer w/o extension"
zuul [Thu, 16 Mar 2017 04:03:52 +0000 (23:03 -0500)]
Merge "res/res_pjsip_refer: call xfer w/o extension"

4 years agoMerge "app_queue: Handle the caller being redirected out of a queue bridge"
zuul [Thu, 16 Mar 2017 01:30:55 +0000 (20:30 -0500)]
Merge "app_queue: Handle the caller being redirected out of a queue bridge"

4 years agoMerge "funcs/func_devstate: Remove new line in Device field of during module load"
zuul [Thu, 16 Mar 2017 01:13:17 +0000 (20:13 -0500)]
Merge "funcs/func_devstate: Remove new line in Device field of during module load"

4 years agoMerge "pbx.c: Fix crash from malformed exten pattern."
zuul [Thu, 16 Mar 2017 00:14:08 +0000 (19:14 -0500)]
Merge "pbx.c: Fix crash from malformed exten pattern."

4 years agoMerge "res_pjsip_endpoint_identifier_ip: Don't output error if no header_match."
zuul [Thu, 16 Mar 2017 00:01:40 +0000 (19:01 -0500)]
Merge "res_pjsip_endpoint_identifier_ip: Don't output error if no header_match."