asterisk/asterisk.git
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: 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 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 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

2 years agochan_iax2: Reload of iax peer results in loss of host address/port
Richard Begg [Tue, 14 Mar 2017 21:51:41 +0000 (08:51 +1100)]
chan_iax2: Reload of iax peer results in loss of host address/port

When using a non-dynamic peer address, build_peer() invalidates the
peer address structure by setting the address family to unspecified.
However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup()
will not amend the peer address if the cache is still valid, resulting
in peer connectivity failures.
To fix this, we call ast_dnsmgr_refresh() instead.

ASTERISK-26865

Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082

2 years agoconfigure: Don't use the progress bar with curl when downloading to stdout
Matt Jordan [Tue, 14 Mar 2017 20:12:28 +0000 (15:12 -0500)]
configure: Don't use the progress bar with curl when downloading to stdout

In some scenarios, such as when there may not be a terminal (such as
inside a Docker container), curl will apparently direct the progress bar
to stdout. This can cause extra data to be appended to a file curl'd
down to stdout, resulting in md5 verification failures.

This patch removes the progress bar, and tells curl to download the file
silently.

ASTERISK-26872 #close

Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c

2 years agoMerge "chan_pjsip: Don't assume a session will have a channel."
zuul [Tue, 14 Mar 2017 19:07:51 +0000 (14:07 -0500)]
Merge "chan_pjsip: Don't assume a session will have a channel."

2 years agoRFC sdp: Initial SDP creation
George Joseph [Thu, 2 Mar 2017 23:11:06 +0000 (16:11 -0700)]
RFC sdp: Initial SDP creation

* Added additional fields to ast_sdp_options.
* Re-organized ast_sdp.
* Updated field names to correspond to RFC4566 terminology.
* Created allocs/frees for SDP children.
* Created getters/setters for SDP children where appropriate.
* Added ast_sdp_create_from_state.
* Refactored res_sdp_translator_pjmedia for changes.

Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48

2 years agoMerge "chan_sip: Call not cancelled after receiving a 422 response"
Joshua Colp [Tue, 14 Mar 2017 16:47:30 +0000 (11:47 -0500)]
Merge "chan_sip: Call not cancelled after receiving a 422 response"

2 years agoconfigs/samples/hep.conf.sample: Clarify how the HEP stack works
Matt Jordan [Tue, 14 Mar 2017 14:55:06 +0000 (09:55 -0500)]
configs/samples/hep.conf.sample: Clarify how the HEP stack works

This patch updates the documenation in hep.conf.sample to better specify
how the various HEP modules interact.

ASTERISK-26717 #close

Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124

2 years agofuncs/func_devstate: Remove new line in Device field of during module load
Matt Jordan [Tue, 14 Mar 2017 14:59:48 +0000 (09:59 -0500)]
funcs/func_devstate: Remove new line in Device field of during module load

During module loading of func_devstate, Asterisk emits the current
device state of all Custom device states currently stored in the AstDB.
This was erroneously including a new line character ('\n') to the end of
the device state, causing two new lines to be emitted in
DeviceStateChange AMI events.

Note that this only happened for those device state changes that
occurred during startup. Regular device state changes for Custom device
states are handled elsewhere, and did not have the newline.

ASTERISK-26643 #close
Reported by: Roman Bedros
Tested by: Matt Jordan
patches:
  ami_devstate.diff uploaded by Roman Bedros (License 6842)

Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93

2 years agomain/stasis_cache: Demote the ERROR message when removing a nonexistent item
Matt Jordan [Tue, 14 Mar 2017 14:37:34 +0000 (09:37 -0500)]
main/stasis_cache: Demote the ERROR message when removing a nonexistent item

This patch demotes the ERROR message that is displayed when a
nonexistent item is removed from the Stasis cache. The genesis of this
demotion is due to chan_sip's realtime peers and their interaction with
Asterisk's core ast_endpoint code, but ostensibly it could happen from
other channel drivers as well.

Since Mark Michelson already did an excellent job of explaining on this
issue, it is quoted here for posterity:

"Internally, when a realtime peer is retrieved, Asterisk creates an
ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
destroyed as well. Part of the destruction of the ast_endpoint involves
clearing the Stasis cache of all information about that endpoint. The
problem here is that the act of creating the ast_endpoint is not enough
to actually put any information in the Stasis cache. Instead, something
has to happen, such as a state change, in order for the Stasis cache to
have any information about that endpoint. When a device registers,
chan_sip creates an ast_endpoint structure, processes the REGISTER, and
then destroys the ast_endpoint. When the ast_endpoint is destroyed,
there is nothing to destroy in the Stasis cache, so an error message is
emitted. When you use rtcachefriends, ast_endpoint structures persist
for the lifetime of the module and so you do not see this error
message."

ASTERISK-25237 #close

Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70

2 years agores_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue
Matt Jordan [Wed, 8 Mar 2017 18:39:20 +0000 (12:39 -0600)]
res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue

Tabs > spaces. Always.

Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1

2 years agochan_pjsip: Don't assume a session will have a channel.
Joshua Colp [Sun, 12 Mar 2017 14:21:16 +0000 (14:21 +0000)]
chan_pjsip: Don't assume a session will have a channel.

When querying for PJSIP specific information using the dialplan
function CHANNEL() it is possible that the underlying session
will no longer have a channel associated with it. This is
most likely to occur when the RTCP HEP module attempts to get
the channel name. If this happens then a crash will occur.

This change just adds a check that the channel exists on the
session before querying it.

ASTERISK-26857

Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01

2 years agopjproject_bundled: Reduce the need for rebuilds
George Joseph [Sat, 11 Mar 2017 02:29:04 +0000 (19:29 -0700)]
pjproject_bundled:  Reduce the need for rebuilds

Bundled pjproject should now only rebuild if one of the menuselect
"Compiler Flags" options changes.

Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43

2 years agoMerge "pjsip/cli_commands: pjsip show channelstats shows wrong codec"
Joshua Colp [Fri, 10 Mar 2017 22:02:08 +0000 (16:02 -0600)]
Merge "pjsip/cli_commands: pjsip show channelstats shows wrong codec"

2 years agoMerge "res_musiconhold: moh general section is a class and issues warning"
zuul [Fri, 10 Mar 2017 00:32:03 +0000 (18:32 -0600)]
Merge "res_musiconhold: moh general section is a class and issues warning"

2 years agoMerge "media_cache: Prefer ast_file_is_readable() over access()"
Joshua Colp [Thu, 9 Mar 2017 22:10:25 +0000 (16:10 -0600)]
Merge "media_cache: Prefer ast_file_is_readable() over access()"

2 years agopjsip/cli_commands: pjsip show channelstats shows wrong codec
Daniel Journo [Sun, 5 Mar 2017 21:26:07 +0000 (21:26 +0000)]
pjsip/cli_commands: pjsip show channelstats shows wrong codec

* cli_commands.c Fixed CLI output

ASTERISK-26822 #close

Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01

2 years agoMerge "pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel"
Joshua Colp [Thu, 9 Mar 2017 20:29:02 +0000 (14:29 -0600)]
Merge "pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel"

2 years agores_musiconhold: moh general section is a class and issues warning
Daniel Journo [Wed, 8 Mar 2017 20:29:16 +0000 (20:29 +0000)]
res_musiconhold: moh general section is a class and issues warning

* res_musiconhold.c: Ensure the general section is not treated as
a moh class.

ASTERISK-26353 #close

Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d

2 years agomedia_cache: Prefer ast_file_is_readable() over access()
Sean Bright [Wed, 8 Mar 2017 23:08:52 +0000 (18:08 -0500)]
media_cache: Prefer ast_file_is_readable() over access()

Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def

2 years agopbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel
Sean Bright [Tue, 7 Mar 2017 12:25:25 +0000 (07:25 -0500)]
pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel

Set a variable on the channel that indicates which attempt number we
are currently performing to allow for attempt-specific behavior.

ASTERISK-26568 #close
Reported by: Roman Shubovich

Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89

2 years agores_pjsip_transport_websocket: Add support for IPv6.
Joshua Colp [Tue, 7 Mar 2017 13:37:52 +0000 (13:37 +0000)]
res_pjsip_transport_websocket: Add support for IPv6.

This change adds a PJSIP patch (which has been contributed upstream)
to allow the registration of IPv6 transport types.

Using this the res_pjsip_transport_websocket module now registers
an IPv6 Websocket transport and uses it for the corresponding
traffic.

ASTERISK-26685

Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647

2 years agoapp_voicemail: Cannot set fromstring on a per-mailbox basis
Daniel Journo [Wed, 8 Mar 2017 14:16:29 +0000 (14:16 +0000)]
app_voicemail: Cannot set fromstring on a per-mailbox basis

* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.

ASTERISK-24562 #close

Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe

2 years agoMerge "res_http_websocket: Fix faulty read logic."
zuul [Wed, 8 Mar 2017 16:05:19 +0000 (10:05 -0600)]
Merge "res_http_websocket: Fix faulty read logic."

2 years agoMerge "pbx_spool: Gracefully handle long lines in call files"
zuul [Tue, 7 Mar 2017 23:54:29 +0000 (17:54 -0600)]
Merge "pbx_spool: Gracefully handle long lines in call files"

2 years agores_http_websocket: Fix faulty read logic.
Mark Michelson [Tue, 7 Mar 2017 19:38:17 +0000 (13:38 -0600)]
res_http_websocket: Fix faulty read logic.

When doing some WebRTC testing, I found that the websocket would
disconnect whenever I attempted to place a call into Asterisk. After
looking into it, I pinpointed the problem to be due to the iostreams
change being merged in.

Under certain circumstances, a call to ast_iostream_read() can return a
negative value. However, in this circumstance, the websocket code was
treating this negative return as if it were a partial read from the
websocket. The expected length would get adjusted by this negative
value, resulting in the expected length being too large.

This patch simply adds an if check to be sure that we are only updating
the expected length of a read when the return from a read is positive.

ASTERISK-26842 #close
Reported by Mark Michelson

Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab

2 years agochan_sip: Call not cancelled after receiving a 422 response
Jean Aunis [Tue, 7 Mar 2017 14:12:48 +0000 (15:12 +0100)]
chan_sip: Call not cancelled after receiving a 422 response

When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.

ASTERISK-26841

Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099

2 years agocore: Add stream topology changing primitives with tests.
Joshua Colp [Tue, 7 Mar 2017 11:22:18 +0000 (11:22 +0000)]
core: Add stream topology changing primitives with tests.

This change adds a few things to facilitate stream topology changing:

1. Control frame types have been added for use by the channel driver
to notify the application that the channel wants to change the stream
topology or that a stream topology change has been accepted. They are
also used by the indicate interface to the channel that the application
uses to indicate it wants to do the same.

2. Legacy behavior has been adopted in ast_read() such that if a
channel requests a stream topology change it is denied automatically
and the current stream topology is preserved if the application is
not capable of handling streams.

Tests have also been written which confirm the multistream and
non-multistream behavior.

ASTERISK-26839

Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9

2 years agoSaynumber is trying to get "and" from "digits/" subfolder
Daniel Journo [Mon, 6 Mar 2017 21:54:23 +0000 (21:54 +0000)]
Saynumber is trying to get "and" from "digits/" subfolder

* say.c Changed 'digits/and' to 'vm-and' for en_GB

ASTERISK-26598 #close

Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe

2 years agopbx_spool: Gracefully handle long lines in call files
Sean Bright [Mon, 6 Mar 2017 19:15:45 +0000 (14:15 -0500)]
pbx_spool: Gracefully handle long lines in call files

Per the linked issue, we aren't checking the buffer filled by fgets()
to determine if it contains a newline, so we will fail to correctly
parse the trailing portion of a long line.

This patch increases the buffer size from 256 to 1024, and skips any
line that exceeds that length, logging a warning in the process.

ASTERISK-17067 #close
Reported by: Dave Olszewski

Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0

2 years agocore: Cleanup ast_get_hint() usage.
Richard Mudgett [Fri, 3 Mar 2017 03:27:00 +0000 (21:27 -0600)]
core: Cleanup ast_get_hint() usage.

* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension.  Ran into this when
developing a testsuite test.  The AMI event ExtensionStatus came out with
the hint header value containing garbage.  The AMI event PresenceStatus
also had the same issue.

* manager.c:action_extensionstate() no need to completely initialize the
hint[].  Only initialize the first element.

* pbx.c:ast_add_hint() Remove unnecessary assignment.

* chan_sip.c: Eliminate an unneeded hint[] local variable.  We only care
about the return value of ast_get_hint() there.

Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b

2 years agoMerge "res_pjsip WebRTC/websockets: Fix usage of WS vs WSS."
Joshua Colp [Wed, 1 Mar 2017 23:24:43 +0000 (17:24 -0600)]
Merge "res_pjsip WebRTC/websockets: Fix usage of WS vs WSS."

2 years agoMerge "stream: Unit tests for stream read and tweaks framework"
Joshua Colp [Wed, 1 Mar 2017 20:58:45 +0000 (14:58 -0600)]
Merge "stream: Unit tests for stream read and tweaks framework"

2 years agoMerge "res_config_pgsql: Make 'require' return consistent with other backends"
Joshua Colp [Wed, 1 Mar 2017 17:24:22 +0000 (11:24 -0600)]
Merge "res_config_pgsql: Make 'require' return consistent with other backends"

2 years agores_pjsip WebRTC/websockets: Fix usage of WS vs WSS.
Jørgen H [Thu, 16 Feb 2017 10:22:47 +0000 (10:22 +0000)]
res_pjsip WebRTC/websockets: Fix usage of WS vs WSS.

According to the RFC[1] WSS should only be used in the Via header
for secure Websockets.

* Use WSS in Via for secure transport.

* Only register one transport with the WS name because it would be
ambiguous.  Outgoing requests may try to find the transport by name and
pjproject only finds the first one registered.  This may mess up unsecure
websockets but the impact should be minimal.  Firefox and Chrome do not
support anything other than secure websockets anymore.

* Added and updated some debug messages concerning websockets.

* security_events.c: Relax case restriction when determining security
transport type.

* The res_pjsip_nat module has been updated to not touch the transport
on Websocket originating messages.

[1] https://tools.ietf.org/html/rfc7118

ASTERISK-26796 #close

Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12

2 years agostream: Unit tests for stream read and tweaks framework
George Joseph [Fri, 24 Feb 2017 21:30:33 +0000 (14:30 -0700)]
stream: Unit tests for stream read and tweaks framework

* Removed the AST_CHAN_TP_MULTISTREAM tech property.  We now rely
  on read_stream being set to indicate a multi stream channel.
* Added ast_channel_is_multistream convenience function.
* Fixed issue where stream and default_stream weren't being set on
  a frame retrieved from the queue.
* Now testing for NULL being returned from the driver's read or
  read_stream callback.
* Fixed issue where the dropnondefault code was crashing on a
  NULL f.
* Now enforcing that if either read_stream or write_stream are
  set when ast_channel_tech_set is called that BOTH are set.
* Added the unit tests.

ASTERISK-26816

Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2

2 years agores_config_pgsql: Make 'require' return consistent with other backends
Sean Bright [Wed, 1 Mar 2017 13:23:55 +0000 (08:23 -0500)]
res_config_pgsql: Make 'require' return consistent with other backends

res_config_pgsql should match the behavior of other realtime backend
drivers so that queue_log can disable adaptive logging.

ASTERISK-25628 #close
Reported by: Dmitry Wagin

Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372

2 years agoSDP: Add initial SDP state machine.
Mark Michelson [Wed, 22 Feb 2017 21:11:29 +0000 (15:11 -0600)]
SDP: Add initial SDP state machine.

This introduces and documents the various states in the state machine.
This also introduces API functions that induce state changes, and places
TODO comments telling what needs to be done in addition to what is
already there. Those TODOs will be replaced with real code in upcoming
changes.

Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed

2 years agoMerge "media_cache: Mark cache entry stale if cache file is removed"
Joshua Colp [Wed, 1 Mar 2017 10:47:59 +0000 (04:47 -0600)]
Merge "media_cache: Mark cache entry stale if cache file is removed"

2 years agoMerge "res_config_pgsql: Release table locks where appropriate"
Joshua Colp [Wed, 1 Mar 2017 01:37:36 +0000 (19:37 -0600)]
Merge "res_config_pgsql: Release table locks where appropriate"

2 years agoMerge "res_pjsip_outbound_registration: Subscribe to network change events"
Joshua Colp [Wed, 1 Mar 2017 01:25:04 +0000 (19:25 -0600)]
Merge "res_pjsip_outbound_registration:  Subscribe to network change events"

2 years agoMerge "build: Warn if asterisk is installed in both 32 and 64 bit sys dirs"
Joshua Colp [Tue, 28 Feb 2017 23:47:46 +0000 (17:47 -0600)]
Merge "build:  Warn if asterisk is installed in both 32 and 64 bit sys dirs"

2 years agoMerge "res_pjsip_pubsub: Remove unneeded endpoint unref"
Joshua Colp [Tue, 28 Feb 2017 23:32:25 +0000 (17:32 -0600)]
Merge "res_pjsip_pubsub:  Remove unneeded endpoint unref"

2 years agoMerge "bridge_native_rtp: Handle case where channel joins already suspended."
zuul [Tue, 28 Feb 2017 23:14:25 +0000 (17:14 -0600)]
Merge "bridge_native_rtp: Handle case where channel joins already suspended."

2 years agomedia_cache: Mark cache entry stale if cache file is removed
Sean Bright [Tue, 28 Feb 2017 19:48:51 +0000 (14:48 -0500)]
media_cache: Mark cache entry stale if cache file is removed

In the event that a cache file is removed out from under us, we should
treat the cache entry as stale and force a refresh.

ASTERISK-26774 #close
Reported by: Igor Gamayunov

Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52

2 years agoMerge "config: Improve documentation and behavior of outbound_proxy option."
Joshua Colp [Tue, 28 Feb 2017 20:44:29 +0000 (14:44 -0600)]
Merge "config: Improve documentation and behavior of outbound_proxy option."

2 years agoMerge "res_pjsip: Fix crash when contact has no status"
Joshua Colp [Tue, 28 Feb 2017 16:24:45 +0000 (10:24 -0600)]
Merge "res_pjsip: Fix crash when contact has no status"

2 years agores_config_pgsql: Release table locks where appropriate
Sean Bright [Tue, 28 Feb 2017 15:41:45 +0000 (10:41 -0500)]
res_config_pgsql: Release table locks where appropriate

The find_table() functions NULL or a locked table pointer. We are
not consistently calling release_table() in failure paths.

Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544

2 years agopjsip.conf.sample: user_agent: not a specific version
Tzafrir Cohen [Tue, 28 Feb 2017 11:41:18 +0000 (13:41 +0200)]
pjsip.conf.sample: user_agent: not a specific version

Use the description of useragent from sip.conf here.

ASTERISK-26825 #close

Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755

2 years agores_pjsip_pubsub: Remove unneeded endpoint unref
George Joseph [Tue, 28 Feb 2017 02:07:06 +0000 (19:07 -0700)]
res_pjsip_pubsub:  Remove unneeded endpoint unref

When a subscription was being recreated and the endpoint wasn't
found, we were trying to unref the endpoint.  This was causing
FRACKs.  Removed the unref.

ASTERISK-26823 #close

Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164

2 years agores_pjsip: Fix crash when contact has no status
Jørgen H [Thu, 16 Feb 2017 10:16:47 +0000 (10:16 +0000)]
res_pjsip: Fix crash when contact has no status

This change fixes an assumption in res_pjsip that a contact will
always have a status. There is a race condition where this is
not true and would crash. The status will now be unknown when
this situation occurs.

ASTERISK-26623 #close

Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5

2 years agores_pjsip_outbound_registration: Subscribe to network change events
George Joseph [Wed, 22 Feb 2017 00:06:00 +0000 (17:06 -0700)]
res_pjsip_outbound_registration:  Subscribe to network change events

Outbound registration now subscribes to network change events
published by res_stun_monitor and refreshes all registrations
when an event happens.

The 'pjsip send (un)register' CLI commands were updated to accept
'*all' as an argument to operate on all registrations.

The 'PJSIP(Un)Register' AMI commands were also updated to
accept '*all'.

ASTERISK-26808 #close

Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25

2 years agobuild: Warn if asterisk is installed in both 32 and 64 bit sys dirs
George Joseph [Mon, 27 Feb 2017 18:25:01 +0000 (11:25 -0700)]
build:  Warn if asterisk is installed in both 32 and 64 bit sys dirs

... and clean them both up on uninstall.

We've fixed the issue where 'make install' was installing to
/usr/lib on 64-bit systems that use /usr/lib64.  Now we need
to clean up the remnants in /usr/lib.

* 'make install' now prints a warning if DESTDIR/ASTLIBDIR
  contains 'lib64' and libasterisk* shared libraries or modules
  are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed
  to 'lib'.

* 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and
  DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'.

ASTERISK-26705

Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f

2 years agobridge_native_rtp: Handle case where channel joins already suspended.
Joshua Colp [Mon, 27 Feb 2017 13:02:38 +0000 (13:02 +0000)]
bridge_native_rtp: Handle case where channel joins already suspended.

The bridge_native_rtp module did not properly handle the case where
a smart bridge operation occurs while a channel is suspended. In this
scenario the module would incorrectly set up local or remote RTP
bridging despite the media having to flow through Asterisk. The remote
endpoint would see two media streams and experience wonky audio.

The module has been changed so that it ensures both channels are
not suspended when performing the native RTP bridging and this
requirement has been documented in the bridge technology.

ASTERISK-26781

Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c

2 years agoMerge "channel: Add ast_read_stream function for reading frames from all streams."
George Joseph [Mon, 27 Feb 2017 14:51:26 +0000 (08:51 -0600)]
Merge "channel: Add ast_read_stream function for reading frames from all streams."

2 years agoMerge "Binaural synthesis (confbridge): DTMF conference management."
zuul [Fri, 24 Feb 2017 23:35:43 +0000 (17:35 -0600)]
Merge "Binaural synthesis (confbridge): DTMF conference management."

2 years agoBinaural synthesis (confbridge): DTMF conference management.
frahaase [Fri, 12 Aug 2016 16:23:40 +0000 (18:23 +0200)]
Binaural synthesis (confbridge): DTMF conference management.

DTMF configuration options for the binaural softmix bridge:
toggle binaural rendering (per channel).

ASTERISK-26292

Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8

2 years agoconfig: Improve documentation and behavior of outbound_proxy option.
Joshua Colp [Fri, 24 Feb 2017 17:49:59 +0000 (17:49 +0000)]
config: Improve documentation and behavior of outbound_proxy option.

This change updates the documentation for the outbound_proxy option
to ensure it is consistently stated that a full SIP URI must be
provided for the option.

The res_pjsip_outbound_registration module has also been changed so
that the provided outbound_proxy value is checked to ensure it is a
URI and if not an error is output stating so.

ASTERISK-26782

Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593

2 years agoMerge "pjproject_bundled: Update for pjproject 2.6"
Joshua Colp [Fri, 24 Feb 2017 18:49:07 +0000 (12:49 -0600)]
Merge "pjproject_bundled: Update for pjproject 2.6"

2 years agoMerge "Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix."
Joshua Colp [Fri, 24 Feb 2017 18:49:00 +0000 (12:49 -0600)]
Merge "Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix."

2 years agoMerge "build: Execute ldconfig to build cache. (take two)"
zuul [Fri, 24 Feb 2017 18:12:13 +0000 (12:12 -0600)]
Merge "build: Execute ldconfig to build cache. (take two)"

2 years agoMerge "channel: Add support for writing to a specific stream."
zuul [Fri, 24 Feb 2017 17:16:13 +0000 (11:16 -0600)]
Merge "channel: Add support for writing to a specific stream."

2 years agochannel: Add ast_read_stream function for reading frames from all streams.
Joshua Colp [Thu, 23 Feb 2017 19:03:15 +0000 (19:03 +0000)]
channel: Add ast_read_stream function for reading frames from all streams.

This change introduces an ast_read_stream function and callback in
the channel technology which allows reading frames from all streams
and not just the default streams.

The stream number has also been added to frames. This is to allow the
case where frames are queued onto the channel instead of being read
directly from the driver.

This change does impose a restriction on reading though: a chain of
frames can only contain frames from the same stream.

ASTERISK-26816

Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce

2 years agopjproject_bundled: Update for pjproject 2.6
George Joseph [Fri, 10 Feb 2017 00:05:01 +0000 (17:05 -0700)]
pjproject_bundled: Update for pjproject 2.6

 * Removed all 2.5.5 functional patches.
 * Updated usages of pj_release_pool to be "safe".
 * Updated configure options to disable webrtc.
 * Updated config_site.h to disable webrtc in pjmedia.
 * Added Richard Mudgett's recent resolver patches.

Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7

2 years agobuild: Execute ldconfig to build cache. (take two)
George Joseph [Thu, 23 Feb 2017 21:49:17 +0000 (14:49 -0700)]
build: Execute ldconfig to build cache. (take two)

On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.  To make matters worse, options were being passed
to ldconfig on both Linux and FreeBSD that actually prevented
the rebuild of the cache.

 * Fedora has a /usr/share/config.site that automatically tells
   autoconf to use /usr/lib64 but CentOS does not. This logic was
   copied to configure.ac and modified so systems like Ubuntu,
   which still use /usr/lib for 64-bit systems, aren't affected.

Now that we have them in the correct directory...

In order for the system loader to find libasteriskssl and
libasteriskpj, one of 3 things has to happen...

  - The linker cache must be rebuilt including the directory
    where the libasterisk* libraries were installed.  Only root
    can rebuild the cache.  This was busted.
  - We have to link the asterisk binary with an rpath pointing
    to the directrory where the libasterisk* libraries were
    installed.  This makes things very complicated and will happen
    over the collective dead bodies of everyone who's had to
    package a distribution with an rpath.
  - Finally, you can start asterisk with LD_LIBRARY_PATH set to the
    directrory where the libasterisk* libraries were installed.

There are no other options. So...

 * The invokation of ldconfig has been moved from main/Makefile
   to ASTTOPDIR/Makefile, the options have been removed, and
   DESTDIR/ASTLIBDIR appended.  If you aren't root, you will be
   warned after the "Asterisk Installation Compete" banner that
   you must re-run 'make install' as root, manually run
   'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with
   LD_LIBRARY_PATH.

ASTERISK-26705

Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982