Walter Doekes [Sun, 10 Sep 2017 11:17:27 +0000 (13:17 +0200)]
res/res_pjsip: Fix localnet checks in pjsip, part 2.
In 45744fc53, I mistakenly broke SDP media address rewriting by
misinterpreting which address was checked in the localnet comparison.
Instead of checking the remote peer address to decide whether we need
media address rewriting, we check our local media address: if it's
local, then we rewrite. This feels awkward, but works and even made
directmedia work properly if you set local_net. (For the record: for
local peers, the SDP media rewrite code is not called, so the
comparison does no harm there.)
Sean Bright [Wed, 6 Sep 2017 15:50:53 +0000 (11:50 -0400)]
app_waitforsilence: Cleanup & don't treat missing frames as 'noise'
* WaitForSilence completes successfully if it receives no media in the
specified timeout, but when acting as WaitForNoise that logic needs
to be reversed.
* Use standard argument parsing macros and add some error checking for
invalid values.
* The documentation indicated that the first argument to both
WaitForSilence and WaitForNoise was required when it was not. Update
the documentation to reflect that.
* Wrap up some behavior in structs to avoid boolean checks all over the
place.
chan_sip: when getting sip pvt return failure if not found
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.
George Joseph [Fri, 1 Sep 2017 10:17:02 +0000 (04:17 -0600)]
stasis/control: Fix possible deadlock with swap channel
If an error occurs during a bridge impart it's possible that
the "bridge_after" callback might try to run before
control_swap_channel_in_bridge has been signalled to continue.
Since control_swap_channel_in_bridge is holding the control lock
and the callback needs it, a deadlock will occur.
* control_swap_channel_in_bridge now only holds the control
lock while it's actually modifying the control structure and
releases it while the bridge impart is running.
* bridge_after_cb is now tolerant of impart failures.
chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE
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.
Alexander Traud [Wed, 6 Sep 2017 08:02:19 +0000 (10:02 +0200)]
res_srtp: Add support for libsrtp2.1.
Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro
SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with
previous versions of libSRTP, this change allows libSRTP 2.1.x as well.
Ben Ford [Tue, 5 Sep 2017 14:35:12 +0000 (09:35 -0500)]
chan_pjsip: Suppress frame warnings.
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.
Sean Bright [Tue, 5 Sep 2017 15:05:48 +0000 (11:05 -0400)]
formats: Restore previous fread() behavior
Some formats are able to handle short reads while others are not, so
restore the previous behavior for the format modules so that we don't
have spurious errors when playing back files.
Walter Doekes [Tue, 5 Sep 2017 14:16:01 +0000 (16:16 +0200)]
res/res_pjsip: Standardize/fix localnet checks across pjsip.
In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was
confusion about whether the transport_state->localnet ACL has ALLOW or
DENY semantics.
For the record: the localnet has DENY semantics, meaning that "not in
the list" means ALLOW, and the local nets are in the list.
Therefore, checks like this look wrong, but are right:
/* See if where we are sending this request is local or not, and if
not that we can get a Contact URI to modify */
if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
ast_debug(5, "Request is being sent to local address, "
"skipping NAT manipulation\n");
(In the list == localnet == DENY == skip NAT manipulation.)
And conversely, other checks that looked right, were wrong.
This change adds two macro's to reduce the confusion and uses those
instead:
res_pjsip_session: Preserve stream name during renegotiation.
Stream names within Asterisk can have meaning so when an externally
initiated renegotiation occurs we need to preserve the name of
the stream if it already exists.
George Joseph [Tue, 5 Sep 2017 10:23:04 +0000 (04:23 -0600)]
res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel
t38_reinvite_response_cb can get called by res_pjsip_session's
session_inv_on_tsx_state_changed in situations where session->channel
is NULL. If it is, the ast_log warning segfaults because it tries
to get the channel name from a NULL channel.
* Check session->channel and print "unknown channel" when it's NULL.
sanitize_tdata was assuming all URIs were SIP URIs so when a non
SIP uri was in the From, To or Contact headers, the unconditional
cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused
a segfault when trying to access uri->other_param.
* Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri)
checks before attempting to cast or use the returned uri.
ASTERISK-27152 Reported-by: Ross Beer
Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f
An admin can configure app_minivm with an externnotify program to be run
when a voicemail is received. The app_minivm application MinivmNotify
uses ast_safe_system() for this purpose which is vulnerable to command
injection since the Caller-ID name and number values given to externnotify
can come from an external untrusted source.
* Add ast_safe_execvp() function. This gives modules the ability to run
external commands with greater safety compared to ast_safe_system().
Specifically when some parameters are filled by untrusted sources the new
function does not allow malicious input to break argument encoding. This
may be of particular concern where CALLERID(name) or CALLERID(num) may be
used as a parameter to a script run by ast_safe_system() which could
potentially allow arbitrary command execution.
* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
instead of ast_safe_system() to avoid command injection.
* Document code injection potential from untrusted data sources for other
shell commands that are under user control.
Joshua Colp [Mon, 22 May 2017 15:36:38 +0000 (15:36 +0000)]
res_rtp_asterisk: Only learn a new source in learn state.
This change moves the logic which learns a new source address
for RTP so it only occurs in the learning state. The learning
state is entered on initial allocation of RTP or if we are
told that the remote address for the media has changed. While
in the learning state if we continue to receive media from
the original source we restart the learning process. It is
only once we receive a sufficient number of RTP packets from
the new source that we will switch to it. Once this is done
the closed state is entered where all packets that do not
originate from the expected source are dropped.
The learning process has also been improved to take into
account the time between received packets so a flood of them
while in the learning state does not cause media to be switched.
Finally RTCP now drops packets which are not for the learned
SSRC if strict RTP is enabled.
Joshua Colp [Wed, 30 Aug 2017 12:28:58 +0000 (12:28 +0000)]
res_rtp_asterisk: Allow remote SSRC to change on an RTP instance.
When SDP renegotiation occurs it is possible for an RTP
instance to be reused for a new stream, resulting in the remote
SSRC changing if it is part of a bundle group. This change
allows this and updates its mapping in the current bundle
group.
Richard Mudgett [Tue, 29 Aug 2017 19:22:15 +0000 (14:22 -0500)]
bridge_native_rtp.c: Fixup native_rtp_framehook()
* Fix framehook to test frame type for control frame.
* Made framehook exit early if frame type is not a control frame.
* Eliminated RAII_VAR in framehook.
* Use switch instead of else-if ladder for control frame handling.
Joshua Colp [Thu, 24 Aug 2017 16:45:08 +0000 (13:45 -0300)]
core: Reduce video update queueing.
A video update frame is used to indicate that a channel
with video negotiated should provide a full frame so the
decoder decoding the stream is able to do so. In situations
where a queue is used to store frames it makes no sense
for the queue to contain multiple video update frames. One
is sufficient to have a full frame be sent.
Sean Bright [Fri, 25 Aug 2017 18:44:35 +0000 (14:44 -0400)]
voicemail: Fix various abuses of mkstemp
mkstemp() returns a unique filename, but appending an extension to that
filename does not guarantee uniqueness. Instead, use mkdtemp() and we
can put whatever extension we want on the files that we create inside
the directory.
In the case of app_minivm, we also now properly clean up any temporary
files that we create.
Sean Bright [Fri, 25 Aug 2017 17:20:16 +0000 (13:20 -0400)]
app_record: Resolve some absolute vs. relative filename bugs
If the Record() application is called with a relative filename that
includes directories, we were not properly creating the intermediate
directories and Record() would fail.
Secondarily, updated the documentation for RECORDED_FILE to mention
that it does not include a filename extension.
Finally, rewrote the '%d' functionality to be a bit more straight
forward and less noisy.
Torrey Searle [Mon, 21 Aug 2017 09:28:52 +0000 (11:28 +0200)]
res/res_pjsip_session: allow SDP answer to be regenerated
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.
Richard Mudgett [Fri, 18 Aug 2017 22:37:12 +0000 (17:37 -0500)]
res_pjsip_session.c: Fix crash when declining an active stream.
If a previously active stream is declined we could crash because the
channel's thread is still using the stream while we are updating the
topology in the serializer thread.
* Defer removing any declined stream's handler until we have blocked the
channel's thread with the channel lock.
Richard Mudgett [Wed, 16 Aug 2017 22:50:18 +0000 (17:50 -0500)]
bridge_channel.c: Fix FRACK when mapping frames to the bridge.
* Add protection checks when mapping streams to the bridge. The channel
and bridge may be in the process of updating the stream mapping when a
media frame comes in so we may not be able to map the frame at the time.
* We need to map the streams to the bridge's stream numbers right before
they are written into the bridge. That way we don't have to keep
locking/unlocking the bridge and we won't have any synchronization
problems before the frames actually go into the bridge.
* Protect the deferred queue with the bridge_channel lock.
Richard Mudgett [Wed, 16 Aug 2017 20:22:04 +0000 (15:22 -0500)]
bridge: Fix softmix bridge deadlock.
* Fix deadlock in
bridge_softmix.c:softmix_bridge_stream_topology_changed() between
bridge_channel and channel locks.
* The new bridge technology topology change callbacks must be called with
the bridge locked. The callback references the bridge channel list, the
bridge technology could change, and the bridge stream mapping is updated.
Michael Kuron [Sun, 20 Aug 2017 13:15:37 +0000 (15:15 +0200)]
res_xmpp: fix inverted return code check in OAuth
fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon
success and -1 if the function is not available.
This commit inverts the return code check so that an error is printed if the
module is not loaded and not if it is loaded.
George Joseph [Wed, 16 Aug 2017 20:43:10 +0000 (14:43 -0600)]
Fix downloader not working with curl
The codec/dpma downloader wasn't handling curl correctly. The logic
that transforms makeopts into a bash-sourceable file wasn't
handling the make 'or' command in DOWNLOAD_TIMEOUT so bash was
looking for an 'or' command.
That logic has been eliminated. Instead of trying to transform
and source makeopts, the downloader now calls a make scriptlet
to print the value of a specific variable. This way, make handles
the ors (or any other make construct that happens to creep into
that file).
Kevin Harwell [Tue, 15 Aug 2017 18:12:10 +0000 (13:12 -0500)]
manager: hook event is not being raised
When the iostream code went in it introduced a conditional that made it so the
hook event was not being raised even if a hook is present. This patch adds a
check to see if a hook is present in astman_append. If so then call into the
send_string function, which in turn raises the even for specified hook.
Also updated the ami hooks unit test, so the test could be automated.
Richard Mudgett [Tue, 15 Aug 2017 20:15:58 +0000 (15:15 -0500)]
configure: Check cache for valid pjproject tarball before downloading.
On a fresh Asterisk source directory, the bundled pjproject tarball is
unconditionally downloaded even if the tarball is already in a specified
cache directory.
* Made check if the pjproject tarball is valid in the cache directory
before downloading the tarball on a fresh source directory.
Richard Mudgett [Tue, 15 Aug 2017 16:14:20 +0000 (11:14 -0500)]
res_pjsip: Fix prune_on_boot to remove only contacts for the host.
* Check that the contact's reg_server matches the host's name before
deleting any prune_on_boot contacts. We don't want to delete reliable
transport contacts made with other servers if the ps_contacts database
table is shared with other servers.
Thanks to Ross Beer for pointing out that the original prune logic would
delete reliable transport contacts from other servers.
Richard Mudgett [Thu, 10 Aug 2017 19:18:01 +0000 (14:18 -0500)]
STUN/netsock2: Fix some valgrind uninitialized memory findings.
* netsock2.c: Test the addr->len member first as it may be the only member
initialized in the struct.
* stun.c:ast_stun_handle_packet(): The combinded[] local array could get
used uninitialized by ast_stun_request(). The uninitialized string gets
copied to another location and could overflow the destination memory
buffer.
These valgrind findings were found for ASTERISK_27150 but are not
necessarily a fix for the issue.
Richard Mudgett [Wed, 2 Aug 2017 23:44:12 +0000 (18:44 -0500)]
res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown.
The fix for the issue is broken up into three parts.
This is part three which handles the client side of REGISTER requests.
The registered contact may no longer be valid on the server when the
transport used is reliable and the connection is broken.
* Re-REGISTER our contact if the reliable transport is broken after
registration completes. We attempt to re-REGISTER immediately to minimize
the time we are unreachable. Time may have already passed between the
connection being broken and the loss being detected.
* Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's
are still correct if an allocation failure happens.
Richard Mudgett [Mon, 31 Jul 2017 19:21:06 +0000 (14:21 -0500)]
res_pjsip: Remove ephemeral registered contacts on transport shutdown.
The fix for the issue is broken up into three parts.
This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled. Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.
* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown. If it is shutdown then the contact must be removed because it
is no longer valid. Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there. Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request. The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.
* Prune any rewrite_contact's registered reliable transport contacts on
boot. The reliable transport no longer exists so the contact is invalid.
* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.
* Made the websocket transport set a unique name since that is what we use
as the ao2 container key. Otherwise, we would not know which transport we
find when one of them shuts down. The names are also used for PJPROJECT
debug logging.
* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event. Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket
transports.
* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.
projects / thirdparty / asterisk.git / log
