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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
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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
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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
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Added missing channel technology read/write stream callback
initialization.
Change-Id: I829043a327d987e0d964485dd3d27964bebbd623
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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
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ASTERISK-26846 #close
Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639
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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
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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
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* cli_commands.c Fixed CLI output
ASTERISK-26822 #close
Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01
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When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.
ASTERISK-26841
Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099
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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
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* 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
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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
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* 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
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When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com),
the user part is stripped down as it would be a trunk with a specified user,
and only the host part is called as a PJSIP endpoint and can't be found.
This is not correct in the case of a multidomain SIP account, so the stripping
after the @ sign is done only if the whole endpoint (in multidomain case
1000@test.com) can't be found.
ASTERISK-26248
Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6
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We shouldn't unlock the channel after starting a snapshot staging because
another thread may interfere and do its own snapshot staging.
* app_dial.c:dial_exec_full() made hold the channel lock while setting up
the outgoing channel staging. Made hold the channel lock after the called
party answers while updating the caller channel staging.
* chan_sip.c:sip_new() completed the channel staging on off-nominal exit.
Also we need to use ast_hangup() instead of ast_channel_unref() at that
location.
* channel.c:__ast_channel_alloc_ap() added a comment about not needing to
complete the channel snapshot staging on off-nominal exit paths.
* rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel
locks while staging the channels for the stats channel variables.
Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a
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The escalator works by creating a set of startup commands in cli.conf
that set up logger channels and issue the debug commands for the
subsystems specified. If asterisk is running when it is executed,
the same commands will be issued to the running instance. The original
cli.conf is saved before any changes are made and can be restored by
executing '$prog --reset'.
The log output will be stored in...
$astlogdir/message.$uniqueid
$astlogdir/debug.$uniqueid
$astlogdir/dtmf.$uniqueid
$astlogdir/fax.$uniqueid
$astlogdir/security.$uniqueid
$astlogdir/pjsip_history.$uniqueid
$astlogdir/sip_history.$uniqueid
Some minor tweaks were made to chan_sip, and res_pjsip_history
so their history output could be send to a log channel as packets
are captured.
A minor tweak was also made to manager so events are output to verbose
when "manager set debug on" is issued.
Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543
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Change-Id: I0a5d56c7dcf327d60f86a4c25a23571733709fd0
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* changes:
res_rtp_asterisk.c: Fix uninitialized memory crash.
chan_rtp.c: Fix uninitialized memory crash.
res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip().
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After a SIP_CODEC_INBOUND in the dialplan, do not continue with cached formats
but remember the joint format. Cached formats contain default parameters,
often create an empty fmtp line. However, a joint format might have passed
format_get_joint(.) in a res_format_attr_* module (like Opus Codec) and
contain the resulting format parameters from a SDP negotiation.
ASTERISK-26691 #close
Change-Id: I35712d98a793d4c3efdd156cec57deab9014b1dc
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The CHANNEL() dialplan function implementation for PJSIP allows
querying of PJSIP specific information. This used the channel
passed in to get the PJSIP session and associated information.
It is possible for this channel to be masqueraded and end
up as a different channel type by the time the information
request is actually acted upon.
This change retrieves the PJSIP session safely and accesses
data from it (including channel). This provides a guarantee
that the session and channel will not be altered when the
request is being acted upon.
ASTERISK-26673
Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6
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unicast_rtp_request() could pass an uninitialized 'us' parameter to
ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us'
parameter may not get initialized. Thus when the code tries to save the
'us' parameter to the local address we could try to copy a ridiculous
sized memory buffer and segfault.
* Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort
the UnicastRTP channel request if it fails.
ASTERISK-26672
Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0
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In some situations TCP threads may become frozen. This creates the
possibility that Asterisk could segfault if they become unfrozen after
chan_sip has been dlclose'd. This reorders the unload_module process to
allow abort if threads do not exit within 5 seconds.
High level order as follows:
1) Unregister from the core to stop new requests.
2) Signal threads to stop
3) Clear config based tables (but do not free the table itself).
4) Verify that threads have shutdown, cancel unload if not.
5) Clean all remaining resources.
ASTERISK-26586
Change-Id: Ie23692041d838fbd35ece61868f4c640960ff882
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Caused by ASTERISK-25494
Change-Id: I1fc408c1a083745ff59da5c4113041bbfce54bcb
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The conditional expressions of the 'if' operators
situated alongside each other are identical.
Change-Id: I652b6dcddb3be007e669a6aa8107edb31a1ddafb
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P is always true. We check it before
Change-Id: Iee61cda002a9f61aee26b9f66c5f9b59e3389efb
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The conditional expressions of the 'if' operators situated
alongside each other are identical.
Change-Id: I2cf7c317b106ec14440c7f1b5dcfbf03639f748a
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RFC says SIP headers look like:
HCOLON = *( SP / HTAB ) ":" SWS
SWS = [LWS] ; sep whitespace
LWS = [*WSP CRLF] 1*WSP ; linear whitespace
WSP = SP / HTAB ; from rfc2234
chan_sip implemented this:
HCOLON = *( LOWCTL / SP ) ":" SWS
LOWCTL = %x00-1F ; CTL without DEL
This discrepancy meant that SIP proxies in front of Asterisk with
chan_sip could pass on unknown headers with \x00-\x1F in them, which
would be treated by Asterisk as a different (known) header. For
example, the "To\x01:" header would gladly be forwarded by some proxies
as irrelevant, but chan_sip would treat it as the relevant "To:" header.
Those relying on a SIP proxy to scrub certain headers could mistakenly
get unexpected and unvalidated data fed to Asterisk.
This change fixes so chan_sip only considers SP/HTAB as valid tokens
before the colon, making it agree on the headers with other speakers of
SIP.
ASTERISK-26433 #close
AST-2016-009
Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b
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The sending codec is switched to the receiving codec and then
is switched back to the best native codec on EVERY receiving RTP packets.
This is because after call of ast_channel_set_rawwriteformat there is call
of ast_set_write_format which calls set_format which sets rawwriteformat
to the best native format.
This patch adds a new function ast_set_write_format_path which set
specific write path on channel and uses this function to switch
the sending codec.
ASTERISK-26603 #close
Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d
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Per RFC 7118 5.2, the SIP URI 'transport' parameter should advertise
'ws' when WebSockets are to be used as the transport. This applies to
both secure and insecure WebSockets.
There were two bugs in Asterisk with respect to this:
(1) The most egregious occurs in res_pjsip. There, we advertise 'ws' for
insecure websockets and 'wss' for secure websockets. While this
would seem to make sense - since 'WS' and 'WSS' are used for the Via
Transport parameter - this is not the case for the SIP URI. This
patch corrects that by registering the secure websockets with
pjproject using the shorthand 'WS', and by returning 'ws' when asked
for the transport parameter. Note that in pjproject, it is perfectly
valid to have multiple transports use the same shorthand.
(2) In chan_sip, we return an upper-case version of the transport 'WS'
instead of 'ws'. Since we should be strict in what we send and
liberal in what we accept (within reason), this patch lower-cases
the transport before appending it to the parameter.
ASTERISK-24330 #close
Reported by: cervajs, Inaki Baz Castillo
Change-Id: Iff77b645f8cc3b7cd35168a6676c26b147f22f42
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If a TCP/TLS connection was pending (not accepted and not timed out) during
unload of chan_sip, Asterisk would segfault when trying to send a signal to
a thread whose thread ID hadn't been recorded yet. This commit fixes that by
recording the thread ID before calling the blocking connect() syscall.
This was a regression introduced by 776a14386a55b5425c7e9617eff8af8b45427144.
The above wasn't enough to fix the segfault, which was now delayed to the
point where connect() timed out. Therefore, it was necessary to also remove
the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be
used to interruput the connect() syscall.
This was a regression introduced by 5d313f51b982a18f7321adcf7c7a4e822d8b2714.
ASTERISK-26586 #close
Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b
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fopencookie/funclose is a non-standard API and should not be used
in portable software. Additionally, the way FILE's fd is used in
non-blocking mode is undefined behaviour and cannot be relied on.
This introduces internal abstraction for io streams, that allows
implementing the desired virtualization of read/write operations
with necessary timeout handling.
ASTERISK-24515 #close
ASTERISK-24517 #close
Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85
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Fix ASTERISK-26565 by adding ast_rtp_instance_stop before
rtp instance destroy for chan_unistim. Also several fixes
for displayed text translation.
Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc
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Correct typo of end-pints to end-points
Re-wrap session timer parameter docs to max 80 chars wide; this
eases reading on terminals with lower resolution, commonly the case
for those with visual impairments.
ASTERISK-26573
Change-Id: I22c94459f4bb6b8a2f6713cfd22e87c32f204e6b
Signed-off-by: C.J. Collier <cjcollier@linuxfoundation.org>
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This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc.
Unfortunately, the aforementioned commit caused a regression (incoming calls
would eventually disconnect). Thus it is being removed.
ASTERISK-26523 #close
ASTERISK-25270
Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d
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