asterisk/asterisk.git
2 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

2 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"

2 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."

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

2 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"

2 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."

2 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"

2 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"

2 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

2 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

2 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"

2 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"

2 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

2 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

2 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

2 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

2 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

2 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

2 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"

2 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"

2 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"

2 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

2 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"

2 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"

2 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"

2 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

2 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"

2 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"

2 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"

2 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."

2 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

2 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

2 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

2 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

2 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

2 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

2 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

2 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

2 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."

2 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."

2 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

2 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"

2 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""

2 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"

2 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

2 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"

2 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

2 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."

2 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"

2 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

2 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

2 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

2 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

2 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

2 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"

2 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

2 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()"

2 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

2 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

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

2 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"

2 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."

2 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."

2 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."

2 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"

2 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"

2 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

2 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

2 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

2 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

2 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

2 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

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

2 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

2 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

2 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"

2 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

2 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

2 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

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

2 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"

2 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"

2 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"

2 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"

2 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."

2 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."

2 years agoautochan/mixmonitor/chanspy: Fix unsafe channel locking and references.
Richard Mudgett [Wed, 15 Mar 2017 18:24:33 +0000 (13:24 -0500)]
autochan/mixmonitor/chanspy: Fix unsafe channel locking and references.

Dereferencing struct ast_autochan.chan without first calling
ast_autochan_channel_lock() is unsafe because the pointer could change at
any time due to a masquerade.  Unfortunately, ast_autochan_channel_lock()
itself uses struct ast_autochan.chan unsafely and can result in a deadlock
if the original channel happens to get destroyed after a masquerade in
addition to the pointer getting changed.

The problem is more likely to happen with v11 and earlier because
masquerades are used to optimize out local channels on those versions.
However, it could still happen on newer versions if the channel is
executing a dialplan application when the channel is transferred or
redirected.  In this situation a masquerade still must be used.

* Added a lock to struct ast_autochan to safely be able to use
ast_autochan.chan while trying to get the channel lock in
ast_autochan_channel_lock().  The locking order is the channel lock then
the autochan lock.  Locking in the other direction requires deadlock
avoidance.

* Fix unsafe ast_autochan.chan usages in app_mixmonitor.c.

* Fix unsafe ast_autochan.chan usages in app_chanspy.c.

* app_chanspy.c: Removed unused autochan parameter from next_channel().

ASTERISK-26867

Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592

2 years agoMerge "core: Add stream topology changing primitives with tests."
zuul [Wed, 15 Mar 2017 22:23:30 +0000 (17:23 -0500)]
Merge "core: Add stream topology changing primitives with tests."

2 years agoAdd rtcp-mux support
Mark Michelson [Tue, 7 Mar 2017 20:13:02 +0000 (14:13 -0600)]
Add rtcp-mux support

This commit adds support for RFC 5761: Multiplexing RTP Data and Control
Packets on a Single Port. Specifically, it enables the feature when
using chan_pjsip.

A new option, "rtcp_mux" has been added to endpoint configuration in
pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with
whatever it communicates with. Asterisk follows the rules set forth in
RFC 5761 with regards to falling back to standard RTCP behavior if the
far end does not indicate support for rtcp-mux.

The lion's share of the changes in this commit are in
res_rtp_asterisk.c. This is because it was pretty much hard wired to
have an RTP and an RTCP transport. The strategy used here is that when
rtcp-mux is enabled, the current RTCP transport and its trappings (such
as DTLS SSL session) are freed, and the RTCP session instead just
mooches off the RTP session. This leads to a lot of specialized if
statements throughout.

ASTERISK-26732 #close
Reported by Dan Jenkins

Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5

2 years agoMerge "res_pjsip_endpoint_identifier_ip: Add an option to match requests by header"
Joshua Colp [Wed, 15 Mar 2017 19:49:13 +0000 (14:49 -0500)]
Merge "res_pjsip_endpoint_identifier_ip: Add an option to match requests by header"

2 years agoMerge "configure: Don't use the progress bar with curl when downloading to stdout"
Joshua Colp [Wed, 15 Mar 2017 18:01:16 +0000 (13:01 -0500)]
Merge "configure: Don't use the progress bar with curl when downloading to stdout"

2 years agores/res_pjsip_refer: call xfer w/o extension
Torrey Searle [Tue, 14 Mar 2017 13:49:54 +0000 (14:49 +0100)]
res/res_pjsip_refer: call xfer w/o extension

When transfering to a URI without an extension, ensure that the
s extension of the dialplan is entered

ASTERISK-26869 #close

Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525

2 years agoapp_queue: Handle the caller being redirected out of a queue bridge
Sean Bright [Thu, 9 Mar 2017 17:05:12 +0000 (12:05 -0500)]
app_queue: Handle the caller being redirected out of a queue bridge

A caller can leave the Queue() application after being bridged with a
member in a few ways:

  * Caller or member hangup
  * Caller is transferred somewhere else (blind or atx)
  * Caller is externally redirected elsewhere

The first 2 scenarios are currently handled by subscribing to stasis
messages, but the 3rd is not explicitly covered. If a caller is
redirected away from the Queue() application, the member who was last
bridged with that caller will remain in an "In use" state until the
caller hangs up.

This patch adds handling of the caller leaving the queue via
redirection. We monitor the caller-member bridge, and if the caller is
the one that leaves, we treat it the same as we would a caller hangup.

ASTERISK-26400 #close
Reported by: Etienne Lessard

Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334

2 years agores_pjsip_endpoint_identifier_ip: Don't output error if no header_match.
Joshua Colp [Wed, 15 Mar 2017 13:44:43 +0000 (13:44 +0000)]
res_pjsip_endpoint_identifier_ip: Don't output error if no header_match.

This change ensures that if no header_match option is set on an
identify an error message is not output stating the option is set
to an invalid value.

ASTERISK-26863

Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a

2 years agores_pjsip_endpoint_identifier_ip: Add an option to match requests by header
Matt Jordan [Tue, 14 Mar 2017 12:50:07 +0000 (07:50 -0500)]
res_pjsip_endpoint_identifier_ip: Add an option to match requests by header

This patch adds a new features to the endpoint identifier module,
'match_header'. When set, inbound requests are matched by a provided SIP
header: value pair. This option works in conjunction with the existing
'match' configuration option, such that if any 'match*' attribute
matches an inbound request, the request is associated with the specified
endpoint.

Since this module now identifies by more than just IP address,
appropriate renaming of the module and/or variables can be done in a
non-release branch.

ASTERISK-26863 #close

Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453
(cherry picked from commit 30f52d79d7fc9ab0b628bef2b61ea515413795a2)

2 years agoMerge "res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue"
George Joseph [Wed, 15 Mar 2017 13:47:36 +0000 (08:47 -0500)]
Merge "res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue"

2 years agoMerge "configs/samples/hep.conf.sample: Clarify how the HEP stack works"
Joshua Colp [Wed, 15 Mar 2017 10:20:50 +0000 (05:20 -0500)]
Merge "configs/samples/hep.conf.sample: Clarify how the HEP stack works"

2 years agoMerge "main/stasis_cache: Demote the ERROR message when removing a nonexistent item"
Joshua Colp [Wed, 15 Mar 2017 10:19:33 +0000 (05:19 -0500)]
Merge "main/stasis_cache: Demote the ERROR message when removing a nonexistent item"

2 years agoMerge "res_pjsip_transport_websocket: Add support for IPv6."
zuul [Wed, 15 Mar 2017 02:22:26 +0000 (21:22 -0500)]
Merge "res_pjsip_transport_websocket: Add support for IPv6."

2 years agopbx.c: Fix crash from malformed exten pattern.
Richard Mudgett [Tue, 14 Mar 2017 21:16:23 +0000 (16:16 -0500)]
pbx.c: Fix crash from malformed exten pattern.

Forgetting to indicate an exten is a pattern can cause a crash if the
"pattern" has a character set range.  e.g., "9999[3-5]" The crash is due
to a buffer overwrite because the '-' exten eye-candy wasn't removed as
expected and overran the allocated space.

The buffer overwrite is fixed two ways in this patch.

1) Fix ext_strncpy() to distinguish between pattern and non-pattern
extens.  Now '-' characters are removed when they are eye-candy and not
when they are part of a pattern character set.  Since the function is
private to pbx.c, the return value now returns the number of bytes written
to the destination buffer instead of the strlen() of the final buffer so
the callers that care don't need to add one.

2) Fix callers to ext_strncpy() to supply the correct available buffer
size of the destination buffer.

ASTERISK-26668

Change-Id: I555d97411140e47e0522684062d174fbe32aa84a