This patch adds regular expression support to make the identify section's
match_header option more useful when attempting to match complex headers
like the 'To' or 'From' headers. The 'From' header has variable
components such as the tag parameter that you cannot predict. To specify
a regular expression put slashes around the regular expression in place of
the header value.
[identify-alice]
type=identify
endpoint=alice
match_header=From: /<sip:alice@127\\.0\\.0\\.1>/
* Added regex support to match_header so you could match a 'To' header
among other complex headers.
Fixed reported crashes when trying to match special headers like 'Contact'.
The identify section's match_header method used code that assumed you were
matching a generic header. Any other type of header could cause a crash
if the header structure variant did not match the generic header enough.
* Made use code that will work for any header type instead of code
specific to generic headers.
Other fixes while in the area:
* Made check all headers of the requested name.
* Added some more sanity checks to the configured identify matching
options when applying the configuration.
ASTERISK-27548
Change-Id: I27dfd4ff5e2259b906640e3c330681b76b4ed1f1
I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.
ASTERISK-27965
Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7
A new option 'suppress_q850_reason_headers' has been added to the
endpoint object. Some devices can't accept multiple Reason headers and
get confused when both 'SIP' and 'Q.850' Reason headers are received.
This option allows the 'Q.850' Reason header to be suppressed.
The default value is 'no'.
ASTERISK-27949
Reported-by: Ross Beer
Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1
pjproject by default currently will follow media forked during an INVITE
on outbound calls if the To tag is different on a subsequent response as
that on an earlier response. We handle this correctly. There have
been reported cases where the To tag is the same but we still need to
follow the media. The pjproject patch in this commit adds the
capability to sip_inv and also adds the capability to control it at
runtime. The original "different tag" behavior was always controllable
at runtime but we never did anything with it and left it to default to
TRUE.
So, along with the pjproject patch, this commit adds options to both the
system and endpoint objects to control the two behaviors, and a small
logic change to session_inv_on_media_update in res_pjsip_session to
control the behavior at the endpoint level.
The default behavior for "different tags" remains the same at TRUE and
the default for "same tag" is FALSE.
Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6
ASTERISK-27936
Reported-by: Ross Beer
Add predial handler support to app_queue. app_dial (ASTERISK_19548) and
app_originate (ASTERISK_26587) have the ability to execute predial
handlers on caller and callee channels. This patch adds predial handlers
to app_queue and uses the same options as Dial and Originate (b and B).
The caller routine gets executed when the caller first enters the queue.
The callee routine gets executed for each queue member when they are about
to be called.
ASTERISK-27912
Change-Id: I5acf5c32587ee008658d12e8a8049eb8fa4d0f24
ConfBridge can now send events to participants via in-dialog MESSAGEs.
All current Confbridge events are supported, such as ConfbridgeJoin,
ConfbridgeLeave, etc. In addition to those events, a new event
ConfbridgeWelcome has been added that will send a list of all
current participants to a new participant.
For all but the ConfbridgeWelcome event, the JSON message contains
information about the bridge, such as its id and name, and information
about the channel that triggered the event such as channel name,
callerid info, mute status, and the MSID labels for their audio and
video tracks. You can use the labels to correlate callerid and mute
status to specific video elements in a webrtc client.
To control this behavior, the following options have been added to
confbridge.conf:
bridge_profile/enable_events: This must be enabled on any bridge where
events are desired.
user_profile/send_events: This must be set for a user profile to send
events. Different user profiles connected to the same bridge can have
different settings. This allows admins to get events but not normal
users for instance.
user_profile/echo_events: In some cases, you might not want the user
triggering the event to get the event sent back to them. To prevent it,
set this to false.
A change was also made to res_pjsip_sdp_rtp to save the generated msid
to the stream so it can be re-used. This allows participant A's video
stream to appear as the same label to all other participants.
Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e
When an AMI client connects, it cannot determine if a user was talking
prior to a transition in the user speaking state (which would generate
a ConfbridgeTalking event). This patch causes app_confbridge to track the
talking state and make this state available via ConfBridgeList.
ASTERISK-27877 #close
Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6
Analog phones dial overlap dialing and it is chan_dahdi's job to read the
numbers. It has three timeout constants that this commit converts to
channel-level configuration options:
* firstdigit_timeout: Default time (ms) to detect first digit
* interdigit_timeout: Default time (ms) to detect following digits
* matchdigit_timeout: Default time (ms) to wait in case of ambiguous
match. This happens when the dialed digits match a number in the current
context but are also the prefix of another number.
Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213
SendText now accepts new channel variables that can be used
to override the To and From display names and set the Content-Type
of a message. Since you can now set Content-Type, other text/*
content types are now valid.
Change-Id: I648b4574478119f95de09d9f08e9595831b02830
Core bridging and, more specifically, bridge_softmix have been
enhanced to relay received frames of type TEXT or TEXT_DATA to all
participants in a softmix bridge. res_pjsip_messaging and
chan_pjsip have been enhanced to take advantage of this so when
res_pjsip_messaging receives an in-dialog MESSAGE message from a
user in a conference call, it's relayed to all other participants
in the call.
res_pjsip_messaging already queues TEXT frames to the channel when
it receives an in-dialog MESSAGE from an endpoint and chan_pjsip
will send an MESSAGE when it gets a TEXT frame. On a normal
point-to-point call, the frames are forwarded between the two
correctly. bridge_softmix was not though so messages weren't
getting forwarded to conference bridge participants. Even if they
were, the bridging code had no way to tell the participants who
sent the message so it would look like it came from the bridge
itself.
* The TEXT frame type doesn't allow storage of any meta data, such
as sender, on the frame so a new TEXT_DATA frame type was added that
uses the new ast_msg_data structure as its payload. A channel
driver can queue a frame of that type when it receives a message
from outside. A channel driver can use it for sending messages
by implementing the new send_text_data channel tech callback and
setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech
properties. If set, the bridging/channel core will use it instead
of the original send_text callback and it will get the ast_msg_data
structure. Channel drivers aren't required to implement this. Even
if a TEXT_DATA enabled driver uses it for incoming messages, an
outgoing channel driver that doesn't will still have it's send_text
callback called with only the message text just as before.
* res_pjsip_messaging now creates a TEXT_DATA frame for incoming
in-dialog messages and sets the "from" to the display name in the
"From" header, or if that's empty, the caller id name from the
channel. This allows the chat client user to set a friendly name
for the chat.
* bridge_softmix now forwards TEXT and TEXT_DATA frames to all
participants (except the sender).
* A new function "ast_sendtext_data" was added to channel which
takes an ast_msg_data structure and calls a channel's
send_text_data callback, or if that's not defined, the original
send_text callback.
* bridge_channel now calls ast_sendtext_data for TEXT_DATA frame
types and ast_sendtext for TEXT frame types.
* chan_pjsip now uses the "from" name in the ast_msg_data structure
(if it exists) to set the "From" header display name on outgoing text
messages.
Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489
This patch adds support to send in-dialog SIP NOTIFY commands on
chan_pjsip channels, similar to the functionality recently added
for chan_sip (ASTERISK_27461).
This extends res_pjsip_notify to allow for in-dialog messages.
ASTERISK-27697
Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29
Add an option to make app_originate not wait for the created channel
to answer.
Change-Id: I7fc2facd77079abc6321f44e8bcd4e39298de2ae
Requested-by: Frederic Steinfels <fst@highdefinition.ch>
Signed-off-by: Russell Bryant <russell@russellbryant.net>
If the two formats on a channel are equal, we don't transcode and since
the generic plc needs slin to work, it doesn't get invoked.
* A new configuration option "genericplc_on_equal_codecs" was added
to the "plc" section of codecs.conf to allow generic packet loss
concealment even if no transcoding was originally needed.
Transcoding via SLIN is forced in this case.
ASTERISK-27743
Change-Id: I0577026a179dea34232e63123254b4e0508378f4
This allows asterisk to be compiled with MALLOC_DEBUG to load modules
built without MALLOC_DEBUG. Now pre-compiled third-party modules will
still work regardless of MALLOC_DEBUG being enabled or not.
Change-Id: Ic07ad80b2c2df894db984cf27b16a69383ce0e10
The pool cache gets in the way of finding use after free errors of memory
pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool
contents are used after free and who freed it.
To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.
Sample pjproject.conf setting:
[startup]
cache_pools=no
* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.
ASTERISK-27704
Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828
Since res_pjsip_transport_management provides several attack
mitigation features, its functionality moved to res_pjsip and
this module has been removed. This way the features will always
be available if res_pjsip is loaded.
ASTERISK-27618
Reported By: Sandro Gauci
Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d
* app_fax (replaced by res_fax).
* res_config_sqlite (replaced by res_config_sqlite3).
* res_monitor (replaced by app_mixmonitor).
This is related to ASTERISK~23657 but does not resolve that ticket.
Resolving that ticket would require complete removal of res_monitor.
ASTERISK-27671 #close
Change-Id: I16a3edd61fc1abd4a7b2e9357693ed663f62dd49
This removes the embedded copy of editline from the Asterisk source
tree, making a system copy of libedit mandatory in Asterisk 16+.
ASTERISK-27634 #close
Change-Id: Iedb64ad92acb78419f3caefedaa2bb7cd2a1a33f
* Made the AMI ConfbridgeList action's ConfbridgeList events output all
the standard channel snapshot headers instead of a few hand-coded channel
snapshot headers. The benefit is that the CallerIDName gets disruptive
characters like CR, LF, Tab, and a few others escaped. However, an empty
CallerIDName is now output as "<unknown>" instead of "<no name>".
ASTERISK-27651
Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977
In an earlier release, inbound registrations on a reliable transport
were pruned on Asterisk restart since the TCP connection would have
been torn down and become unusable when Asterisk stopped. This same
process is now also applied to inbound subscriptions.
Also fixed issues in res_pjsip_registrar where it wasn't handling the
monitoring correctly when multiple registrations came in over the same
transport.
To accomplish this, the pjsip_transport_event feature needed to
be refactored to allow multiple monitors (multiple subcriptions or
registrations from the same endpoint) to exist on the same transport.
Since this changed the API, any external modules that may have used the
transport monitor feature (highly unlikey) will need to be changed.
ASTERISK-27612
Reported by: Ross Beer
Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36
Each time the dial plan is reloaded, a lot of logs like these are generated:
"Added extension 'XXXXX' priority 1 to YYYYYYYYYYY"
This patch changes the log level for those logs.
ASTERISK-27084
Change-Id: I5662902161c50890997ddc56835d4cafb456c529
This patch adds the ability to configure a prompt which will be read
to the "winner" who pressed 1 (or the configured value) and received
the call.
ASTERISK-24372 #close
Change-Id: I6ec1c6c883347f7d1e1f597189544993c8d65272
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
There has been an open issue against cdr_syslog (ASTERISK~14441) about
a race condition for 7.5 years that has never been addressed. Because
this module is effectively unmaintained and currently broken, there is
no sense in keeping it around.
If logging CDRs to syslog is a desirable feature, it would probably be
better to write the logs directly to the syslog server via socket
instead of using the facilities provided by openlog/syslog/closelog.
Doing so would address the race condition referenced in the associated
issue.
Change-Id: Ic77b94cd97f355a9cf5b1d3f3444964a6e0ba5dc
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
Log a message to security events when an INVITE is received to an
invalid extension.
ASTERISK-25869 #close
Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f
A couple of places were setting the status to "UNKNOWN" when qualifies were
being disabled. Instead this should be set to the "CREATED" status that
represents when a contact is given (uri available), but the qualify frequency
is set to zero so we don't know the status.
This patch updates the relevant places with "CREATED". It also updates the
"CREATED" status description (value shown in CLI/AMI/ARI output) to a value
of "NonQualified"/"NonQual" as this description is hopefully less confusing.
ASTERISK-27467
Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89
The media frame cache gets in the way of finding use after free errors of
media frames. Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.
* Added the "cache_media_frames" option to asterisk.conf. Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the frame is
used after free and who freed it. NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.
To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.
Sample asterisk.conf setting:
[options]
cache_media_frames=no
ASTERISK-27413
Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00
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
Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function
instead of correct 'dtmf_features'
ASTERISK-27377 #close
Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930
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
* Mark the module deprecated.
* Disable the module by default.
* Produce a warning the first time a macro is used.
* Note deprecation related options in app_dial and app_queue.
ASTERISK-27350
Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc
When "rewrite_contact" is enabled, the "max_contacts" count option can
block re-registrations because the source port from the endpoint can be
random. When the re-registration is blocked, the endpoint may give up
re-registering and require manual intervention.
* The "remove_existing" option now allows a registration to succeed by
displacing any existing contacts that now exceed the "max_contacts" count.
Any removed contacts are the next to expire. The behaviour change is
beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
than one. The removed contact is likely the old contact created by
"rewrite_contact" that the device is refreshing.
ASTERISK-27192
Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b
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
Add ability to use tokens instead of passwords according to Google OAuth 2.0
protocol.
ASTERISK-27169
Reported by: Andrey Egorov
Tested by: Andrey Egorov
Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db
'--with-pjproject-bundled' is now the default when running
./configure. It can be disabled with '--without-pjproject-bundled'.
To make building without an internet connection easier, a new
./configure option '--with-download-cache' was added that sets
the cache for externals (like pjproject, the codecs and the DPMA),
AND the sounds files. It can also be specified as an environment
variable named "AST_DOWNLOAD_CACHE". The existing
'--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and
'--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable
remain and if specified, will override '--with-downloads-cache'.
ASTERISK-27189
Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce
This adds a way to access information passed along with SIP headers in
a REFER message that initiates a transfer. Headers matching a dialplan
variable GET_TRANSFERRER_DATA in the transferrer channel are added to
a HASH object TRANSFER_DATA to be accessed with functions HASHKEY and HASH.
The variable GET_TRANSFERRER_DATA is interpreted to be a prefix for
headers that should be put into the hash. If not set, no headers are
included. If set to a string (perhaps 'X-' in a typical case), all headers
starting this string are added. Empty string matches all headers.
If there are multiple of the same header, only the latest occurrence in
the REFER message is available in the hash.
Obviously, the variable GET_TRANSFERRER_DATA must be inherited by the
referrer channel, and should be set with the '_' or '__' prefix.
I avoided a specific reference to SIP or REFER, as in my mind the mechanism
can be generalized to other channel techs.
ASTERISK-27162
Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e
Syntax: SIP_HEADERS([prefix])
If the argument is specified, only the headers matching the given prefix
are returned.
The function returns a comma-separated list of SIP header names from an
incoming INVITE message. Multiple headers with the same name are included
in the list only once. The returned list can be iterated over using the
functions POP() and SIP_HEADER().
For example, '${SIP_HEADERS(Co)}' might return the string
'Contact,Content-Length,Content-Type'.
Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional
extended headers sent by a peer.
ASTERISK-27163
Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267
The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to
date.
Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2
Setting this option will cause the Queue application to only announce
the caller's position if it has improved since the last time that we
announced it.
Change-Id: I173a124121422209485b043e2bf784f54242fce6
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis
ASTERISK-27085 #close
Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
By default, when res_musiconhold reloads or unloads, it sends a HUP
signal to custom applications (and all descendants), waits 100ms,
then sends a TERM signal, waits 100ms, then finally sends a KILL
signal. An application which is interacting with an external
device and/or spawns children of its own may not be able to exit
cleanly in the default times, expecially if sent a KILL signal, or
if it's children are getting signals directly from
res_musiconhoild.
* To allow extra time, the 'kill_escalation_delay'
class option can be used to set the number of milliseconds
res_musiconhold waits before escalating kill signals, with the
default being the current 100ms.
* To control to whom the signals are sent, the "kill_method" class
option can be set to "process_group" (the default, existing
behavior), which sends signals to the application and its
descendants directly, or "process" which sends signals only to the
application itself.
Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b
There wasn't any good way to pass options like --host or --build
down to the pjproject configure which makes cross-compiling difficult.
* Added a new PJPROJECT_CONFIGURE_OPTS environment variable which
can be used to pass arbitrary options to pjproject configure.
* Automatically set the pjproject configure --host and --build
options to match those supplied for the asterisk configure.
ASTERISK-27097 #close
Reported-by: Kinsey Moore
Change-Id: I5fa776e110262851173002a26ffe1172e4c35b2e
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
A new global option "imap_poll_logout" was added to specify whether need to
disconnect from the IMAP server after polling of mailboxes.
ASTERISK-27068 #close
Closing IMAP connection after loading mailbox from voicemail.conf
ASTERISK-24052 #close
Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a
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
PJSIP support in Asterisk differs from chan_sip in that it
allows media to be sent as-is without transcoding provided
the codecs were negotiated in the SDP. This is allowed
according to the RFC. Support for this differs quite a lot
though and some endpoints do not handle it well.
This change extends the 'asymmetric_rtp_codec' option to
also cover this case. When set to no (the default) the code
behaves as chan_sip does - the best codec is selected and
we will only ever send that, unless we change what we are
sending if the remote side changes. When set to yes we
will send media as-is without transcoding if the codec
has been negotiated in the SDP.
ASTERISK-26996
Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51
This introduces the ability for PJSIP code to specify filtering flags
when retrieving PJSIP contacts. The first flag for use causes the
query code to only retrieve contacts that are not unreachable. This
change has been leveraged by both the Dial() process and the
PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt
calls to contacts which are not unreachable.
ASTERISK-26281
Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c
This change allows the format of the EAGI audio pipe to be changed by
setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of
the loaded formats.
ASTERISK-26124 #close
Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd
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
This change adds the required logic to allow the SIP
Call-ID to be placed into the HEP RTCP traffic if the
chan_sip module is used. In cases where the option is
enabled but the channel is not either SIP or PJSIP then
the code will fallback to the channel name as done
previously.
Based on the change on Nir's branch at:
team/nirs/hep-chan-sip-support
ASTERISK-26427
Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d
All log messages go to a queue serviced by a single thread
which does all the IO. This setting controls how big that
queue can get (and therefore how much memory is allocated)
before new messages are discarded. The default is 1000.
Should something go bezerk and log tons of messages in a tight
loop, this will prevent memory escalation.
When the limit is reached, a WARNING is logged to that effect
and messages are discarded until the queue is empty again. At
that time another WARNING will be logged with the count of
discarded messages. There's no "low water mark" for this queue
because the logger thread empties the entire queue and processes it
in 1 batch before going back and waiting on the queue again.
Implementing a low water mark would mean additional locking as
the thread processes each message and it's not worth it.
A "test" was added to test_logger.c but since the outcome is
non-deterministic, it's really just a cli command, not a unit
test.
Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1
Added the stun_blacklist option to rtp.conf. Some multihomed servers have
IP interfaces that cannot reach the STUN server specified by stunaddr.
Blacklist those interface subnets from trying to send a STUN packet to
find the external IP address. Attempting to send the STUN packet
needlessly delays processing incoming and outgoing SIP INVITEs because we
will wait for a response that can never come until we give up on the
response. Multiple subnets may be listed.
ASTERISK-26890 #close
Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342
We needed the reason for our reporting when agents pause/unpause all of
their queues at once. This is a small, simple patch that adds a reason
for PAUSEALL and UNPAUSEALL. I have been using it in production for years.
ASTERISK-26920 #close
Change-Id: Ifb3f0d1a0abd5194253d9794023546e1395baf3d
Two new parameters have been added to the pjsip config wizard.
* Setting 'sends_line_with_registrations' to true will cause the wizard
to skip the creation of an identify object to match incoming request
to the endpoint and instead add the line and endpoint parameters to
the outbound registration object.
* Setting 'outbound_proxy' is a shortcut for adding individual
endpoint/outbound_proxy, aor/outbound_proxy and
registration/outbound_proxy parameters.
Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0
(cherry picked from commit a827892ff7)
(cherry picked from commit 27344675be)
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
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
* 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
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
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
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 30f52d79d7)
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
* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.
ASTERISK-24562 #close
Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe
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
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
Original patch by John Covert, slight modifications by me.
ASTERISK-17428 #close
Reported by: John Covert
Patches:
app_voicemail.c.patch (license #5512) patch uploaded by
John Covert
Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6
When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.
This patch adds the 'u' option to Record() to override that behavior.
ASTERISK-18286 #close
Reported by: var
Patches:
app_record-1.8.7.1.diff (license #6184) patch uploaded by var
Change-Id: Ia1cd163483235efe2db05e52f39054288553b957
The 'ari set debug' command has been enhanced to accept 'all' as an
application name. This allows dumping of all apps even if an app
hasn't registered yet. To accomplish this, a new global_debug global
variable was added to res/stasis/app.c and new APIs were added to
set and query the value.
'ari set debug' now displays requests and responses as well as events.
This required refactoring the existing debug code.
* The implementation for 'ari set debug' was moved from stasis/cli.{c,h}
to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted.
* In order to print the body of incoming requests even if a request
failed, the consumption of the body was moved from the ari stubs
to ast_ari_callback in res_ari.c and the moustache templates were
then regenerated. The body is now passed to ast_ari_invoke and then
on to the handlers. This results in code savings since that template
was inserted multiple times into all the stubs.
An additional change was made to the ao2_str_container implementation
to add partial key searching and a sort function. The existing cli
code assumed it was already there when it wasn't so the tab completion
was never working.
Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf
(cherry picked from commit 1d890874f3)
This change implements SRV support for the IP based endpoint
identifier module. All possible addresses through SRV are looked
up and added as matches. If no SRV records are available a
fallback to normal host resolution is done. If an IP address
is provided then no SRV lookup occurs.
This is configured using the "srv_lookups" option on the
identify section and defaults to "yes".
ASTERISK-26693
Change-Id: I6b641e275bf96629320efa8b479737062aed82ac
Adds the ability for extensions to be registered to include filename and
line number so that dialplan show output can show the filename and line
number of a config file responsible for generating a given extension.
This only affects config modules that are written to use the new extension
registering functions. In this patch, that only includes pbx_config, so
extensions registered in extensions.conf and any included extension will
be shown in this manner. Extensions registered in this manner will show
the filename and line number *instead* of the registrar.
ASTERISK-26658 #close
Reported by: Jonathan R. Rose
Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30
The PJSIPShowRegistrationsInbound AMI command was just dumping out
all AORs which was pretty useless and resource heavy since it had
to get all endpoints, then all aors for each endpoint, then all
contacts for each aor.
PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail
events which meets the intended purpose of the other command and has
significantly less overhead. Also, some additional fields that were
added to Contact since the original creation of the ContactStatusDetail
event have been added to the end of the event.
For compatibility purposes, PJSIPShowRegistrationsInbound is left
intact.
ASTERISK-26644 #close
Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a
Increasing the testsuite shutdown timeout before forcibly killing
Asterisk allowed more events to be sent out. Some tests failed as
a result. The tests/channels/pjsip/statsd/registrations failed
because we now get the statsd events that a comment in the test
configuration stated couldn't be intercepted. Unfortunately, we
get a variable number of events because of internal status state
transition races generating redundant statsd events.
We were reporting redundant statsd PJSIP.registrations.state changes
for internal state changes that equated to the same thing publicly.
* Made update_client_state_status() filter out redundant statsd
updates.
ASTERISK-26527
Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646
Use of the new logging is as simple as issuing the new CLI command or
setting the new pjproject.conf option.
Other options that can affect the logging are how you have the pjproject
log levels mapped to Asterisk log types in pjproject.conf and if you have
configured Asterisk to log the DEBUG type messages. Altering the
pjproject.conf level mapping shouldn't be necessary for most installations
as the default mapping is sensible. Configuring Asterisk to log the DEBUG
message type is standard practice for collecting debug information.
* Added CLI "pjproject set log level" command to dynamically adjust the
maximum pjproject log message level.
* Added CLI "pjproject show log level" command to see the currently set
maximum pjproject log message level.
* Added pjproject.conf startup section "log_level" option to set the
initial maximum pjproject log message level so all messages could be
captured from initialization.
* Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into
bundled pjproject. Pjproject will use the currently set run time log
level to determine if a log message is generated just like Asterisk
verbose and debug logging levels.
* In log_forwarder(), made always log enabled and mapped pjproject log
messages. DEBUG mapped log messages are no longer gated by the current
Asterisk debug logging level.
* Removed RAII_VAR() from res_pjproject.c:get_log_level().
ASTERISK-26630 #close
Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992
that requested ability to add callerid into app_originate.
Comments in that issue suggested that it was better solved by
adding an option to gosub prior to originating the call. The
attached patch implements this much like app_dial with two
options one to gosub on the originating channel and one to gosub
on the newly created channel and behaves just like app_dial.
I have tested this patch by adding callerid info to the new
channel and also SIPAddHeader (to e.g. add header to force auto
answer) and confirmed it works. Have also tested both 'exten'
and 'app' versions of app_originate.
Opened by: dkerr
Patch by: dkerr
Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57
In order to not have version number overlap between different versions
of Asterisk, each new major version of Asterisk will mean we also bump
the ARI major version number.
This particular change does NOT introduce any known breaking changes to
ARI.
For discussion relating to this topice, see:
http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html
Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665
In multi-party bridges, Asterisk currently supports two video modes:
* Follow the talker, in which the speaker with the most energy is shown
to all participants but the speaker, and the speaker sees the
previous video source
* Explicitly set video sources, in which all participants see a locked
video source
Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.
This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
Removes any explicit video source, and sets the video mode to talk
detection
ASTERISK-26595 #close
Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621
Richard Mudgett
Corey Farrell
Joshua Colp
George Joseph
Jenkins2
Kristian F. Høgh
William McCall
Tzafrir Cohen
Nathan Bruning
Russell Bryant
Sean Bright
krells
ghjm
Sungtae Kim
Rodrigo Ramírez Norambuena
Kevin Harwell
Igor Goncharovskiy
Thomas Sevestre
Andrey Egorov
kkm
Torrey Searle
Alexei Gradinari
Troy Bowman
zuul
Richard Begg
Mark Michelson
Matt Jordan
Daniel Journo
Jørgen H
Sebastian Gutierrez
Jonathan R. Rose
David Kerr