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ASTERISK-27603
Change-Id: I65c587534c0ae364f063d68da1bed40bb3d5e8aa
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The installation script and the new configure option --with-pjproject-bundled
aimed to accomplish the same. However, the installation script was out of
date. Users should go for the maintained configure option, or the Wiki.
ASTERISK-24598
Change-Id: Icbf4b562f81f7c05bd24a3805bd46c0beb4ebd44
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ASTERISK-27555
Change-Id: Ieb41b0cbf968af12882b39454b819ebb48b9ea46
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This re-enables the script ./contrib/scripts/install_prereq on Fedora 22 and
newer, and on RHEL/CentOS when the option strict=1 was set for yum install.
ASTERISK-27598
Reported by: Hunter Stevens, Said Masoud
Change-Id: I40f9517122aaa6906e8fc0962b4b8008dfddb368
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The type=identify endpoint identification method can match by IP address
and by SIP header. However, the SIP header matching has limited
usefulness because you cannot specify the SIP header matching priority
relative to the IP address matching. All the matching happens at the same
priority and the order of evaluating the identify sections is
indeterminate. e.g., If you had two type=identify sections where one
matches by IP address for endpoint alice and the other matches by SIP
header for endpoint bob then you couldn't predict which endpoint is
matched when a request comes in that matches both.
* Extract the SIP header matching criteria into its own "header" endpoint
identification method so the user can specify the relative priority of the
SIP header and the IP address matching criteria in the global
endpoint_identifier_order option. The "ip" endpoint identification method
now only matches by IP address.
ASTERISK-27491
Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095
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ASTERISK-27555
Change-Id: I0818b6e42631be1b69237e2b41d3415275693e53
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Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
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configs/samples/minivm.conf.sample contains invalid UTF-8, but that
appears to be intentional.
Change-Id: I7b1e0d332f3380fd0425962a3c9c55f9b200c8cc
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This patch adds the ability to set the wrapuptime on the queue member
config.
When the option is set the wrapuptime on the queue member is used instead
of the queue's wrapuptime.
ASTERISK-27483 #close
Change-Id: I11c85809537f974eb44dc5bbf82bcedd8a458902
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The OUTPUTDIR environment variable can now be set either in the
environment itself or in ast_debug_tools.conf. If set, it's used
for all work products instead of /tmp.
Also added the --tarball-config option that includes the contents
of /etc/asterisk when either --tarball-coredumps or --tarball-results
are used.
Change-Id: I66b2553319df61caea5b313d084f51978f730b4c
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This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.
Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.
ASTERISK-27395
Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4
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Since Asterisk 13.17, libSRTP 2.x is supported. Therefore, its latest version
is installed again via the script install_prereq.
ASTERISK-27356
Change-Id: I13125839a79052356469e41edacbebff0a937d39
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Adds an extra option, --asterisk-bin=<path> to ast_coredumper. If
provided, the binary given to gdb will be the parameter, rather than
asterisk from the PATH.
ASTERISK-27380 #close
Change-Id: I25f5b91eb75059b0fb2f142e468c26b283b0a9f3
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The ps_endpoints table was missing the bundle column
introduced with the bundle feature in
commit 065c3005ad92.
ASTERISK-27374 #close
Change-Id: Ic900f4f2c20f64b99ea898d50f5c0a7117472d46
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When the identify_by option on an endpoint is set to ip it will
only be identified using the res_pjsip_endpoint_identifier_ip module.
This ensures that it is not mistakenly matched using the username of
the From header. To ensure behavior has not changed the default has
been changed to "username,ip" for the identify_by option.
ASTERISK-27206
Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd
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The --tarball-coredump option now creates a gzipped tarball of
coredumps processed, their results txt files and copies of
/etc/os-release, /usr/sbin/asterisk, /usr/lib(64)/libasterisk* and
/usr/lib(64)/asterisk as those files are needed to properly examine
the coredump. The file will be named
/tmp/asterisk.<timestamp>.coredumps.tar.gz or
/tmp/asterisk-<uniqueid>.coredumps.tar.gz if --tarball-uniqueid was
specified.
Added dumps of *_siginfo to the top of the txt files so you can
tell what signal was invoked.
Change-Id: Ib9ee6d83592d4b1bc90cb3419a05376a88d1ded9
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when 'all' is specified in an allow or disallow section, it should erase
all values from the inverse section in the default config. E.G.
allow=all should erase any deny values from default config &
vice-versa
ASTERISK-27333 #close
Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6
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add mappings for udptl redundancy, rtptimeout, and debug flags
Change-Id: Ie73cf5c83c05dee01eb9624ede76c1a30225d73a
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A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to
receive unsolicited MWI NOTIFY requests and make them available to
other modules via the stasis message bus.
res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request"
that parses a simple-message-summary body and, if
endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state
with the voice-message counts from the message.
Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c
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The downgrade function was missing "_v2" at the end of the
alter column type.
Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e
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MS-SQL has no native Enum-type support and therefore
needs to work with constraints.
Since these constraints need unique names the suggested approach
referenced in the following alembic documentation has been applied:
http://bit.ly/2x9r8pb
ASTERISK-27255 #close
Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95
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Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db
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Added include for postgresql ENUM type and
redefined values in the same way as in the
other migration scripts.
ASTERISK-27254 #close
Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa
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The ps_endpoints table was missing the dtls_fingerprint column
introduced with commit adba2a8d7fd.
ASTERISK-27168 #close
Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd
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The fix for the issue is broken up into three parts.
This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled. Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.
* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown. If it is shutdown then the contact must be removed because it
is no longer valid. Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there. Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request. The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.
* Prune any rewrite_contact's registered reliable transport contacts on
boot. The reliable transport no longer exists so the contact is invalid.
* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.
* Made the websocket transport set a unique name since that is what we use
as the ao2 container key. Otherwise, we would not know which transport we
find when one of them shuts down. The names are also used for PJPROJECT
debug logging.
* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event. Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket
transports.
* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.
ASTERISK-27147
Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4
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When the "webrtc" option was added in res_pjsip it was not added to the alembic
scripts. This patch adds the option for alembic.
Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of
an OPT_BOOL_T so if this field is ever written to a database it will write out
the correct value.
ASTERISK-27119 #close
Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b
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Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5
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AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0
Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago
Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7
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Also add new corosync packages to install_prereq.
Reported by Travis Ryan in #asterisk-dev
Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db
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Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187
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The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated. This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.
ASTERISK-27066 #close
Change-Id: Id185b11e84afd9191a2f269e8443019047765e91
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The stream topology (list of streams and order) is now stored with the
configured PJSIP endpoints and used during the negotiation process.
Media negotiation state information has been changed to be stored
in a separate object. Two of these objects exist at any one time
on a session. The active media state information is what was previously
negotiated and the pending media state information is what the
media state will become if negotiation succeeds. Streams and other
state information is stored in this object using the index (or
position) of each individual stream for easy lookup.
The ability for a media type handler to specify a callback for
writing has been added as well as the ability to add file
descriptors with a callback which is invoked when data is available
to be read on them. This allows media logic to live outside of
the chan_pjsip module.
Direct media has been changed so that only the first audio and
video stream are directly connected. In the future once the RTP
engine glue API has been updated to know about streams each individual
stream can be directly connected as appropriate.
Media negotiation itself will currently answer all the provided streams
on an offer within configured limits and on an offer will use the
topology created as a result of the disallow/allow codec lines.
If a stream has been removed or declined we will now mark it as such
within the resulting SDP.
Applications can now also request that the stream topology change.
If we are told to do so we will limit any provided formats to the ones
configured on the endpoint and send a re-invite with the new topology.
Two new configuration options have also been added to PJSIP endpoints:
max_audio_streams: determines the maximum number of audio streams to
offer/accept from an endpoint. Defaults to 1.
max_video_streams: determines the maximum number of video streams to
offer/accept from an endpoint. Defaults to 1.
ASTERISK-27076
Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7
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This change adds support for socket activation of certain SOCK_STREAM
listeners in Asterisk:
* AMI / AMI over TLS
* CLI
* HTTP / HTTPS
Example systemd units are provided. This support extends to any socket
which is initialized using ast_tcptls_server_start, so any unknown
modules using this function will support socket activation.
Asterisk continues to function as normal if socket activation is not
enabled or if systemd development headers are not available during
build.
ASTERISK-27063 #close
Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d
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This option was added to control whether to notify dialog-info state
'early' or 'confirmed' on Ringing when already INUSE.
The value "yes" is useful for some SIP phones (Cisco SPA)
to be able to indicate and pick up ringing devices.
ASTERISK-26919 #close
Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711
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This option was added to turn off notifying the progress details
on Blind Transfer. If this option is not set then the chan_pjsip
will send NOTIFY "200 OK" immediately after "202 Accepted".
Some SIP phones like Mitel/Aastra or Snom keep the line busy until
receive "200 OK".
ASTERISK-26333 #close
Change-Id: Id606fbff2e02e967c02138457badc399144720f2
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This change adds an Alembic migration which adds a
ps_resource_list table that can contain resource_list
RLS configuration objects.
ASTERISK-26929
Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05
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This change adds database tables for the PUBLISH support so it
can be configured using realtime. A minor fix to the
res_pjsip_publish_asterisk module was done so that it read the
sorcery configuration from the correct section. Finally the
sample configuration files have been updated.
ASTERISK-26928
Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952
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Change-Id: I744c986da4a38aeff8c00837eb89de7841fbc86c
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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
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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
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