chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant
it was not updating HANGUPCAUSE for 4XX responses. If the remote end
sent a "180 Ringing", then a "486 Busy", the hangup cause was left at
"180 Normal Clearing".
* Removed chan_pjsip_incoming_response from the original session
supplement (which was handling only "AFTER MEDIA") and added it to a
new session supplement which accepts both "BEFORE_MEDIA" and
"AFTER_MEDIA".
* Also cleaned up some cleanup code in load module.
ASTERISK-27902
Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a
This function originally was used in chan_sip to enable some simplifying
assumptions and eventually was copy and pasted into res_pjsip_logger and
res_hep. Since it's replicated in three places, it's probably best to
move it into the public netsock2 API for these modules to use.
Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04
This fixes build warnings found by GCC 8. In some cases format
truncation is intentional so the warning is just suppressed.
ASTERISK-27824 #close
Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84
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
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
clang 6.0 warned about this. Beside that, this change removes the used variable
'desc'.
ASTERISK-27808
Change-Id: Ia26bdcc0a562c058151814511cfcf70ecafa595b
ast_sip_push_task_synchronous() did not necessarily execute the passed in
task under the specified serializer. If the current thread is any
registered pjsip thread then it would execute the task immediately instead
of under the specified serializer. Reentrancy issues could result if the
task does not execute with the right serializer.
The original reason ast_sip_push_task_synchronous() checked to see if the
current thread was a registered pjsip thread was because of a deadlock
with masquerades and the channel technology's fixup callback
(ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356)
involving call pickups avoided the original deadlock situation entirely.
The PJSIP channel technology's fixup callback no longer needed to call
ast_sip_push_task_synchronous().
However, there are a few places where this unexpected behavior is still
required to avoid deadlocks. The pjsip monitor thread executes callbacks
that do calls to ast_sip_push_task_synchronous() that would deadlock if
the task were actually pushed to the specified serializer. I ran into one
dealing with the pubsub subscriptions where an ao2 destructor called
ast_sip_push_task_synchronous().
* Split ast_sip_push_task_synchronous() into
ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer().
ast_sip_push_task_wait_servant() has the old behavior of
ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has
the new behavior where the task is always executed by the specified
serializer or a picked serializer if one is not passed in. Both functions
behave the same if the current thread is not a SIP servant.
* Redirected ast_sip_push_task_synchronous() to
ast_sip_push_task_wait_servant() to preserve API for released branches.
ASTERISK_26806
Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3
There is a problem when an INVITE-with-Replaces transfer targets a channel
in a ConfBridge. The transfer will unconditionally swap out the
ConfBridge channel. Unfortunately, the ConfBridge state will not be aware
of this change. Unexpected behavior will happen as a result since
ConfBridge channels currently can only be replaced by a masquerade and not
normal bridge channel moves.
* We just need to pretend that the channel isn't in a bridge (like other
transfer methods already do) so the transfer channel will masquerade into
the ConfBridge channel.
Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82
This change allows chan_pjsip to be given an AST_FRAME_RTCP
containing REMB feedback and pass it to res_rtp_asterisk.
Once res_rtp_asterisk receives the frame a REMB RTCP feedback
packet is constructed with the appropriate contents and sent
to the remote endpoint.
ASTERISK-27776
Change-Id: Ic53f821c1560d8924907ad82c4d9c0bc322b38cd
Given the below call scenario:
A -> Ast1 -> B
C <- Ast2 <- B
1) A calls B through Ast1
2) B calls C through Ast2
3) B transfers A to C
When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to
send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of
the channel between Ast1 and Ast2.
Channel ref leaks are easily seen in the CLI "core show channels" output.
The leaked channels appear in the output but you can do nothing with them
and they never go away unless you restart Asterisk.
* Properly account for the channel refs when imparting a channel into a
bridge when handling an INVITE with replaces in handle_invite_replaces().
The ast_bridge_impart() function steals a channel ref but the code didn't
account for how many refs were held by the code at the time and which ref
was stolen.
* Eliminated RAII_VAR in handle_invite_replaces().
ASTERISK-27740
Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4
This change extends the existing AST_FRAME_RTCP frame type to be
able to contain additional RTCP message types, such as feedback
messages. The payload type is contained in the subclass which allows
knowing what is in the frame itself.
The RTCP feedback message type is now handled and REMB[1] messages
are raised with their containing information.
This also fixes a bug where all feedback messages were triggering
video updates instead of just FIR and FUR.
Finally RTCP frames are now passed up through the Asterisk core to
what is handling the channel, mapped appropriately in the case of
bridging, and written to an outgoing stream. Since RTCP frames are
on a per-stream basis this is only done on multistream capable
channels.
[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
ASTERISK-27758
ASTERISK-26366
Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e
In the script ./configure, AST_EXT_LIB_CHECK checks for external libraries. Some
libraries do not specify all their dependencies and require additional shared
libraries. In AST_EXT_LIB_CHECK, this is the fifth parameter. However, if a
library is specified there, it must exist on the platform, because ./configure
tries to compile/link/execute a small app using those statements. For example,
the library libdl.so is Linux specific and does not exist on BSD-like platforms.
Furthermore, no supported platform/version was found, which still (ever?)
requires those additional libraries. Therefore, they were simply removed.
Finally, this change adds the error code ESTRPIPE to the channel driver
chan_alsa for those platforms which lack it, again for example NetBSD.
ASTERISK-27720
Change-Id: I3b21f2135f6cbfac7590ccdc2df753257f426e0b
* acl (named_acl.c)
* cdr
* cel
* ccss
* dnsmgr
* dsp
* enum
* extconfig (config.c)
* features
* http
* indications
* logger
* manager
* plc
* sounds
* udptl
These modules are now loaded at appropriate time by the module loader.
Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so
the module loader will abort startup on failure of these modules.
Some of these modules are still initialized or shutdown from outside the
module loader. logger.c is initialized very early and shutdown very
late, manager.c is initialized by the module loader but is shutdown by
the Asterisk core (too much uses it without holding references).
Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f
Checking option_debug directly is incorrect as it ignores file/module
specific debug settings. This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.
Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0
The "ptime" SDP parameter received in a SIP response was not honoured.
Moreover, in the abscence of this "ptime" parameter, locally configured
framing was lost during response processing.
This patch systematically stores the framing information in the
ast_rtp_codecs structure, taking it from the response or from the
configuration as appropriate.
ASTERISK-27674
Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c
When constructing a dialog-info+xml NOTIFY message a ringing channel
is found if the state is ringing and further information is placed into
the message. Due to the migration to the Stasis message bus this did
not always work as expected.
This change raises a second ringing event in such a way to guarantee
that the event is received by chan_sip and another lookup is done to
find the ringing channel.
ASTERISK-24488
Change-Id: I547a458fc59721c918cb48be060cbfc3c88bcf9c
Check if initreq data string exists before using it when processing a
CANCEL request.
ASTERISK-27666
Change-Id: Id1d0f0fa4ec94e81b332b2973d93e5a14bb4cc97
It seems that the ALSA backend of PortAudio doesn't know how to both
read and write at the same time by adding a per-device mutex.
FIXME: currently only a draft version. Need to either auto-detect
we work with the ALSA backend or add an extra configuration option
to use this mutex.
ASTERISK-27426 #close
Change-Id: I635eacee45f5413faa18f5a3b606af03b926dacb
This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
This patch fix chan_unistim hold functions to correctly support
hold function in different states possible in case of multiple lines
established on the phone
ASTERISK-26596 #close
Change-Id: Ib1e04e482e7c8939607a42d7fddacc07e26e14d4
* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
Per RFC 5245, the foundation specified with an ICE candidate can be up
to 32 characters but we are only allowing for 31.
ASTERISK-27498 #close
Reported by: Michele Prà
Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf
This is the old ASTOBJ macro's which are no longer used except by the
deprecated netsock.c. Move it to the chan_iax2 include folder so it
does not get used elsewhere.
Change-Id: I7e4ae96678b36b9f41d3cae14b167f110eb5d349
Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh
ASTERISK-24198 #close
Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
In change_redirecting_information variables we use ast_strlen_zero to
see if a value should be saved. In the case where the value is not NULL
but is a zero length string we leaked.
handle_response_subscribe leaked a reference to the ccss monitor
instance.
Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f
Some variables are set and never changed, making them constant. This
means that code in the 'false' block of the conditional is unreachable.
In chan_skinny and res_config_ldap I used preprocessor directive `#if 0`
as I'm unsure if the unreachable code could be enabled in the future.
Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059
chan_console supports multiple devices but the CLI only works on a
single device. 'console set active' selects this device.
Sadly that CLI picks the wrong command-line parameter and will only
work for a device called 'active'.
ASTERISK-27490 #close
Change-Id: I2f0e5fe63db19845bee862575b739360797dc73d
This moves netsock.c / netsock.h to the chan_iax2 module. netsock.h has
been marked deprecated since 13.0.0, chan_iax2 is the only remaining
user.
Change-Id: I28c6578043bac18de5ea608e136acec4f83d5dd3
Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and
disable_multi_domain=no results in a misleading empty endpoint name
message. The message should say the endpoint was not found.
* Added missing endpoint not found message.
* Added more information to the empty endpoint name msgs if available.
* Eliminated RAII_VAR in request().
Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4
Log a message to security events when an INVITE is received to an
invalid extension.
ASTERISK-25869 #close
Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f
Remove nearly all use of regex from ACO users. Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
callers use simple prefix based regex. I haven't decided the best
way to fix this in both 13/15 and master.
Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b
Stripping the DNID in a SIP dial string can result in attempting to call
the argument parsing macros on an empty string, causing a crash.
ASTERISK-26131 #close
Reported by: Dwayne Hubbard
Patches:
dw-asterisk-master-dnid-crash.patch (license #6257) patch
uploaded by Dwayne Hubbard
Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
ASTERISK-27478
Change-Id: I512e60d1f162395c946451becb37af3333337b33
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
Previously, peers connected via TCP (or TLS) were matched by ignoring their
source port. One cannot say anything when protocol:IP:port match, yes (see
<http://stackoverflow.com/q/3329641>). However, when the ports do not match, the
peers do not match as well.
This change allows two peers connected to an Asterisk server via TCP (or TLS)
behind a NAT (= same source IP address) to be differentiated via their port as
well.
ASTERISK-27457
Reported by: Stephane Chazelas
Change-Id: Id190428bf1d931f2dbfd4b293f53ff8f20d98efa
chan_skinny creates a new thread for each new session. In trying
to be a good cleanup citizen, the threads are joinable and the
unload_module function does a pthread_cancel() and a pthread_join()
on any sessions that are active at that time. This has an
unintended side effect though. Since you can call pthread_join on a
thread that's already terminated, pthreads keeps the thread's
storage around until you explicitly call pthread_join (or
pthread_detach()). Since only the module_unload function was
calling pthread_join, and even then only on the ones active at the
tme, the storage for every thread/session ever created sticks
around until asterisk exits.
* A thread can detach itself so the session_destroy() function
now calls pthread_detach() just before it frees the session
memory allocation. The module_unload function still takes care
of the ones that are still active should the module be unloaded.
ASTERISK-27452
Reported by: Juan Sacco
Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd
(cherry picked from commit 8f5dff543e)
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
When chan_sip receives a SUBSCRIBE request with no "Expires" header it
processes the request as an unsubscribe. This is incorrect, per RFC3264
when the "Expires" header is missing a default expiry should be used.
ASTERISK-18140
Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5
When sip.conf contains 'sipdebug=yes' it is impossible to disable it
using CLI 'sip set debug off'. This corrects the output of that CLI
command to instruct the user to turn sipdebug off in the configuration
file.
ASTERISK-23462 #close
Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318
The sys/sysmacros.h include file does not exist in BSD systems and
is not required to build this module there.
Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section
already exist I moved that include line inside it's #else branch.
ASTERISK-27343 #close
Change-Id: Ibfb64f4e9a0ce8b6eda7a7695cfe57916f175dc1
chan_vpb was trying to use sizeof(*p->play_dtmf), where
p->play_dtmf is defined as char[16], to get the length of the array
but since p->play_dtmf is an actual array, sizeof(*p->play_dtmf)
returns the size of the first array element, which is 1. gcc7
validly complains because the context in which it's used could
cause an out-of-bounds condition.
Change-Id: If9c4bfdb6b02fa72d39e0c09bf88900663c000ba
Currently privacy requests are only granted if the Privacy header
value is exactly "id" (defined in RFC 3325). It ignores any other
possible value (or a combination there of). This patch reverses the
logic from testing for "id" to grant privacy, to testing for "none" and
granting privacy for any other value. "none" must not be used in
combination with any other value (RFC 3323 section 4.2).
ASTERISK-27284 #close
Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56
Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.
This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.
ASTERISK-27277
Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07
chan_pjsip_indicate was missing a case for the recently added
AST_CONTROL_STREAM_TOPOLOGY_CHANGED condition and was returning an
error and causing the call to be hung up instead of just ignoring
it.
ASTERISK-27260
Reported by: Daniel Heckl
Change-Id: I4fecbb00a0b8a853da85155065c1a6bddf235e80
Provide a way to get the contents of the the Request URI from the initial SIP
INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}")
ASTERISK-27278
Reported by: David J. Pryke
Tested by: David J. Pryke
Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e
Multicast/Unicast RTP do not use SDP so we need to use a format that
cleanly maps to one of the static RTP payload types. Without this
change, an Originate to a Multicast or Unicast channel without a format
specified would produce no audio on the receiving device.
ASTERISK-21399 #close
Reported by: Tzafrir Cohen
Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3
In handle_request_invite, when processing a pickup, a call
is made to get_sip_pvt_from_replaces to locate the pvt for
the subscription. The pvt is assumed to be valid when zero
is returned indicating no error, and is dereferenced which
can cause a crash if it was not found.
This change checks the not found case and returns -1 which
allows the calling code to fail appropriately.
ASTERISK-27217 #close
Reported-by: Bryan Walters
Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612
If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE
to both parties to set up media path directly between the endpoints.
In this reINVITE msg SDP origin line (o=) contains IP address of endpoint
instead of IP of asterisk. This behavior violates RFC3264, sec 8:
"When issuing an offer that modifies the session,
the "o=" line of the new SDP MUST be identical to that in the
previous SDP, except that the version in the origin field MUST
increment by one from the previous SDP."
This patch assures IP address of Asterisk is always sent in
SDP origin line.
ASTERISK-17540
Reported by: saghul
Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e
When rtp_keepalive is on for a PJSIP endpoint dialing to another
Asterisk instance also using PJSIP, Asterisk will continue to print
warning messages about not being able to send frames of a certain
type. This suppresses that warning message.
Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67
Create local_tag and remote_tag in CHANNEL info to get tag from From and
To headers of a SIP dialog.
ASTERISK-27220
Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524
If an SDP answer hasn't been sent yet, it's legal to change it.
This is required for PJSIP_DTMF_MODE to work correctly, and can
also have use in the future for updating codecs too.
ASTERISK-27209 #close
Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1
Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not. If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed
ASTERISK-26745 #close
Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f
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
* chan_sip: channel in test_sip_rtpqos_1.
* test_config: config hook, config info and global config holder.
* test_core_format: format in format_attribute_set_without_interface.
* test_stream: unneeded frame duplication.
* test_taskprocessor: task_data.
Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31
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
GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
ASTERISK-27156 #close
Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
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
On every reload of chan_iax2 module, MWI subscription was added, which
results in additional taskprocessors being accumulated over time.
This commit fixes it by making sure we check for existing subscription
first.
This was verified with 'core show taskprocessors' CLI command.
ASTERISK-27122 #close
Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9
This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:
rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.
Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.
ASTERISK-27119 #close
Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
BUNDLE is a specification used in WebRTC to allow multiple
streams to use the same underlying transport. This reduces
the number of ICE and DTLS negotiations that has to occur
to 1 normally.
This change implements this by adding support for it to
the RTP SDP module in PJSIP. BUNDLE can be turned on using
the "bundle" option and on an offer we will offer to
bundle streams together. On an answer we will accept any
bundle groups provided. Once accepted each stream is bundled
to another RTP instance for transport.
For the res_rtp_asterisk changes the ability to bundle
an RTP instance to another based on the SSRC received
from the remote side has been added. For outgoing traffic
if an RTP instance is bundled to another we will use the
other RTP instance for any transport related things. For
incoming traffic received from the transport instance we
look up the correct instance based on the SSRC and use it
for any non-transport related data.
ASTERISK-27118
Change-Id: I96c0920b9f9aca7382256484765a239017973c11
This API was not actively maintained, was not added to new modules
(such as res_pjsip), and there exist better alternatives to acquire the
same information, such as the ARI.
Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83
When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was
added in any case, because of a local Boolean-negation error of the return value
of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was
still always added with tlsenable=yes, because the domains were not compared
just on the address but also on the port – and TLS is always on a different port
than UDP/TCP.
ASTERISK-27106
Change-Id: I14fe9e319e238320b094016980445ef3a5b3337c
Because of a copy-and-paste error when the struct ast_sockaddr changed,
tlsbindaddr was not added, when sip.conf contained autodomain=yes; see
"show sip domains" on the command-line interface (CLI) of Asterisk.
ASTERISK-27106
Change-Id: I3d0957150017c223136968ef1266f275d0d6695e
When connected_line_method is "invite", we're supposed to determine
if the client can support UPDATE and if it can, send UPDATE instead
of INVITE to avoid the SDP renegotiation. Not only was pjproject
not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing
that invite_tsx wasn't NULL which isn't always the case.
* Updated chan_pjsip/update_connected_line_information to drop the
requirement that invite_tsx isn't NULL.
* Submitted patch to pjproject sip_inv.c that sets the
PJSIP_INV_SUPPORT_UPDATE flag correctly.
* Updated pjsip.conf.sample to clarify what happens when "invite"
is specified.
ASTERISK-27095
Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560
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
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
The construction of the returned string assumed incorrectly that the
supplied buffer would always be initialized as an empty string. If it is
not an empty string we could overrun the supplied buffer by the length of
the non-empty buffer string plus one. It is also theoreticaly possible
for the supplied buffer to be overrun by a string terminator during a read
operation even if the supplied buffer is an empty string.
* Fix the assumption that the supplied buffer would already be an empty
string. The buffer is not guaranteed to contain an empty string by all
possible callers.
* Fix string terminator buffer overrun potential.
Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9
The chan_pjsip module uses a calculation approach for
determining device state. This means that in situations
where we would expect device state to change we need to
tell the core to query. A scenario that was missed is
when early media was signaled.
This change adds the notification for the core to
query device state when we are told that early media
is being provided.
ASTERISK-27039
Change-Id: Iafebfd152894966344ff2e950a3cee9f59a3eb6f
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
If we are offered or are offering RTCP-MUX, don't consider RTCP ICE
candidates. This confuses certain browsers (current Firefox for
example) and causes intial audio setup delays.
ASTERISK-26982 #close
Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91
The while(1) loop in skinny_session wasn't checking for EOF so
a packet that was longer than a header but still truncated
would spin the while loop infinitely. Not only does this
permanently tie up a thread and drive a core to 100% utilization,
the call of ast_log() in such a tight loop eats all available
process memory.
Added poll with timeout to top of read loop
ASTERISK-26940 #close
Reported-by: Sandro Gauci
Change-Id: I2ce65f3c5cb24b4943a9f75b64d545a1e2cd2898
If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed. This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.
* ast_sip_session_terminate was modified to explicitly call the
cleanup tasks and unreference session if the invite state is NULL
AND invite_tsx is NULL (meaning we never sent a transaction).
* chan_pjsip/hangup was modified to bump session before it calls
ast_sip_session_terminate to insure that session stays valid
while it does its own cleanup.
* Added test events to session_destructor for a future testsuite
test.
ASTERISK-26908 #close
Reported-by: Richard Mudgett
Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9
For outgoing TCP connections, Asterisk uses the first IP address of the
interface instead of the IP address we asked him to bind to.
ASTERISK-26922 #close
Reported-by: Ksenia
Change-Id: I43c71ca89211dbf1838e5bcdb9be8d06d98e54eb
Some equipments may send a re-INVITE containing an SDP in the final ACK
request. If this happens in the context of direct media, the remote end
should be updated with a re-INVITE.
This patch queues an "update RTP peer" frame to trigger the re-INVITE,
instead of the "source change" frame wich was used previously.
ASTERISK-26951
Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6
This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compact_headers=yes via the file pjsip.conf.
ASTERISK-26932 #close
Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689
In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
SIP user-agents indicate which protocol extensions are allowed in headers
like Supported and Required. Such protocol extensions are Session Timers
(RFC 4028) for example. Session Timers are supported since Mantis-10665.
Since ASTERISK-21721, not only the first but multiple Supported/Required
headers in a message are parsed. In that change, an existing variable was
re-used within a newly added do-loop. Currently, at the end of that loop,
that variable is an empty string always. Previously, that variable was used
within log output. However, the log output was not changed.
ASTERISK-26915 #close
Change-Id: I09315f31b4d78fb214bb2a9fb6c0f5e143eae990
POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.
Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.
Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48
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
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
When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.
ASTERISK-26841
Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099
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
* 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
There is difference exists in behaviour of char type on x86 and ARM.
On x86 by default char variable type means signed char, but in ARM
unsigned char used. This make binary calculations and negative values
works wrong on ARM.
This patch change type of char variables used for store negative
values and binary calculations to signed char.
ASTERISK-26714
Change-Id: Id78716dee9568a58419d4ef63c038affc3dfc7ab
* app_minivm: Use built-in completion facilities to complete optional
arguments.
* app_voicemail: Use built-in completion facilities to complete
optional arguments.
* app_confbridge: Add missing colons after 'Usage' text.
* chan_alsa: Use built-in completion facilities to complete optional
arguments.
* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'
* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'
* func_odbc: Correct completions for 'odbc read' and 'odbc write'
* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.
* main/bridge: Correct completions for 'bridge kick.'
* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.
* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'
* main/pbx_app: Remove redundant completions for 'core show
applications.'
* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'
* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.
Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
Jenkins2
Alexander Traud
George Joseph
Joshua Colp
Matthew Fredrickson
Corey Farrell
Tzafrir Cohen
Richard Mudgett
Jean Aunis
Oron Peled
Igor Goncharovsky
Sean Bright
Yasuhiko Kamata
Guido Falsi
Daniel Tryba
David J. Pryke
Scott Griepentrog
Vitezslav Novy
Ben Ford
Andre Nazario
Torrey Searle
kkm
Sergej Kasumovic
Kevin Harwell
Mark Michelson
Thierry Magnien
zuul
Richard Begg
Daniel Journo