Jaco Kroon [Tue, 3 Dec 2019 18:27:38 +0000 (20:27 +0200)]
netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr.
ast_addressfamily_to_sockaddrize will determine the size that's
required, and ast_sockaddr_from_sockaddr then wraps this new function
and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without
knowing the address family) into the ast_sockaddr structure.
Kevin Harwell [Wed, 8 Jan 2020 19:54:44 +0000 (13:54 -0600)]
app_agent_pool: Update XML docs for AgentLogin
This patch fixes some wrongly formatted documentation for the AgentLogin
application. A couple of "see also" links should contain only the function
name, and no parameters.
George Joseph [Wed, 8 Jan 2020 18:11:26 +0000 (11:11 -0700)]
CI: Update buildAsterisk.sh to do a "make full"
If you do a "make all" when building Asterisk the xml documentation
produced will be missing certain AMI events where their
documentation is located not at the top of the c source file but
embedded further down next to the event's manager_event()
registration call. See main/manager_mwi.c for an example.
"make full" does produce the correct documentation so we're changing
it in the build script. A separate commit/issue will address the
problem with "make all".
Joshua C. Colp [Mon, 6 Jan 2020 15:02:54 +0000 (15:02 +0000)]
res_pjsip_pubsub: Add ability to persist generator state information.
Some body generators, such as dialog-info+xml, require storing state
information which is then conveyed in the NOTIFY request itself. Up
until now there was no way for such body generators to persist this
information.
Two new API calls have been added to allow body generators to set and
get persisted data. This data is persisted out alongside the normal
persistence information and allows the body generator to restore
state information or to simply use this for normal storage of state.
State is stored in the form of JSON and it is up to the body
generator to interpret this as needed.
The dialog-info+xml body generator has been updated to take advantage
of this to persist the version number.
Sean Bright [Tue, 24 Dec 2019 15:16:23 +0000 (10:16 -0500)]
res_pjsip_endpoint_identifier_ip.c: Add port matching support
Adds source port matching support when IP matching is used:
[example]
type = identify
match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444
If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.
George Joseph [Mon, 30 Dec 2019 17:04:23 +0000 (10:04 -0700)]
sig_pri: Fix deadlock caused by sig_pri_queue_hangup
The change to add setting hangupsource to sig_pri_queue_hangup()
made in https://gerrit.asterisk.org/c/asterisk/+/12857 casued
deadlocks when a hangup request was received from the core at the
same time a hanguprequest was received from the remote end via the
D channel.
Although the PRI's channel private structure was being unlocked
before setting the hangupsource, the PRI's own lock was still being
held during the process. If channel actions were also coming from
the core, a deadlock on the PRI could result. This deadlock could
then escalate to the entire DAHDI subsystem via DAHDI's global
interface list lock, especially if someone used the PRI CLI commands.
Fix:
* We now unlock the PRI as well as the PRI's channel private
structure before setting the hangupsource, then relock both
afterwards.
Richard Mudgett [Mon, 30 Dec 2019 19:13:46 +0000 (13:13 -0600)]
app_chanisavail.c: Simplify dialplan using ChanIsAvail.
Dialplan has to be careful about passing an empty device list or empty
positions in the list. As a result, dialplan has to check for these
conditions before using ChanIsAvail. Simplify dialplan by making
ChanIsAvail handle these conditions gracefully.
* Made tolerate empty positions in the device list.
* Simplified the code and eliminated some unnecessary indention.
George Joseph [Thu, 2 Jan 2020 20:25:33 +0000 (13:25 -0700)]
stasis.c: Use correct topic name in stasis_topic_pool_delete_topic
When a topic is created for an object, its name is only
<object>:<uniqueid>
For example:
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
When a topic is added to a pool, its name has the pool's topic
name prepended. For example:
bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
The topic_pool_entry's name however, is only what was passed
in to stasis_topic_pool_get_topic which is
bridge:cb68b3a8-fce7-4738-8a17-d7847562f020
That's actually correct because the entry is qualified by the
pool that's in.
When you're ready to delete the entry from the pool, you retrieve
the tropic name from the object but since it now has the pool's
topic name prepended, it won't be found in the pool container.
Fix:
* Modified stasis_topic_pool_delete_topic() to skip past the
pool topic's name, if it was prepended to the topic name,
before searching the container for a pool entry.
Richard Mudgett [Mon, 30 Dec 2019 04:38:05 +0000 (22:38 -0600)]
app_dial.c: Simplify dialplan using Dial.
Dialplan has to be careful about passing an empty destination list or
empty positions in the list. As a result, dialplan has to check for
these conditions before using Dial. Simplify dialplan by making Dial
handle these conditions gracefully.
* Made tolerate empty positions in the dialed device list.
* Reduced some message log levels from notice to verbose.
Richard Mudgett [Mon, 30 Dec 2019 02:41:30 +0000 (20:41 -0600)]
app_page.c: Simplify dialplan using Page.
Dialplan has to be careful about passing an empty destination list or
empty positions in the list. As a result, dialplan has to check for
these conditions before using Page. Simplify dialplan by making Page
handle these conditions gracefully.
* Made tolerate empty positions in the paged device list.
* Reduced some warnings associated with the 's' option to verbose
messages. The warning level for those messages really serves no purpose
as that is why the 's' option exists.
Richard Mudgett [Mon, 30 Dec 2019 00:36:54 +0000 (18:36 -0600)]
features.c: Make Bridge application tolerate unspecified channel.
The Bridge application was inconsistent if the channel to bridge with is
not specified. If no parameters are given then a warning is issued and
the current channel is hung up. If options are given but no channel is
specified then a warning is issued and the current channel is not hung up.
* Made the Bridge application give a verbose message instead of a warning
if the channel to bridge with is not specified and made not hang up the
current channel. As a result dialplan no longer needs to check if a
channel name is passed before calling Bridge and simply needs to check the
BRIDGERESULT channel variable instead. This is something you likely want
your dialplan to do anyway.
* Fixed up L() option warning message. It is up to the caller to
determine if the channel is hung up because of the warning. Dial() hangs
up the current channel while Bridge() does not.
Sean Bright [Sun, 5 Jan 2020 16:00:46 +0000 (11:00 -0500)]
res_pjsip_config_wizard: Fix change detection for wizard settings
ast_sorcery_changeset_create() is not commutative and will fail to detect
differences between two variable lists depending on what changed, so switch to
ast_variable_lists_match().
Sean Bright [Tue, 26 Nov 2019 19:24:10 +0000 (14:24 -0500)]
websocket: Consider pending SSL data when waiting for socket input
When TLS is in use, checking the readiness of the underlying FD is insufficient
for determining if there is data available to be read. So before polling the
FD, check if there is any buffered data in the TLS layer and use that first.
Kevin Reeves [Thu, 21 Nov 2019 18:48:42 +0000 (12:48 -0600)]
main/file.c: Limit media cache usage to remote files.
When testing for the existance of a file, the media cache is searched even if
the file has no chance of being in it. This can cause performance issues
as the media cache size increases.
As a result, calls to applications like Read and Playback using local files
must scan through the media cache before playing. Under load and with a
large cache, this can delay the playback of those files.
This patch updates the function that checks for the existance of a file to
only consult the media cache database if the requested file is a remote path.
It introduces a new is_remote_path() function in main/file.c.
Frederic LE FOLL [Wed, 27 Nov 2019 17:34:24 +0000 (18:34 +0100)]
app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.
Temporary channel lifespan is very short and CDR deactivation request
through ast_cdr_set_property() may happen when CDR is not available
yet. Use CDR_PROP() dialplan function instead, it will first wait
for pending CDR insertion requests to be processed.
Joshua C. Colp [Mon, 16 Dec 2019 12:35:31 +0000 (08:35 -0400)]
configure: Add check for MySQL client bool and my_bool type usage.
Instead of trying to use the defined MySQL client version from the
header use a configure check to determine whether the bool or my_bool
type should be used for defining a boolean.
Joshua C. Colp [Thu, 12 Dec 2019 00:03:46 +0000 (00:03 +0000)]
confbridge: Add support for specifying maximum sample rate.
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.
Joshua C. Colp [Mon, 16 Dec 2019 11:23:07 +0000 (07:23 -0400)]
res_pjsip_session: Set stream state on created streams for incoming SDP.
A previous review, 13174, made a change whereby on an incoming offer SDP
the pending topology was initialized to the configured. This caused a problem
for bundle with WebRTC where bundle could reference a stream that did not
actually exist if the configuration had both audio and video but the
offer SDP only contained audio.
This change undoes that review and instead fixes the original problem it
sought to solve by setting the state of created streams based on the
contents of the offer SDP. This way the stream state is not inactive
until negotiation later completes.
made it so a T.38 Gateway tries to negotiate with both sides by sending T.38
negotiation request to both endpoints supported T.38 versus the previous
behavior of forwarding negotiation to the "other" channel once a preamble
was detected.
This had the unfortunate side effect of breaking some setups. Specifically
ones that set the max datagram option on an endpoint configuration (configured
max datagram was not propagated since Asterisk now initiates negotiations).
This patch adds a configuration option, "negotiate_both", that when enabled
makes it so Asterisk initiates the negotiation requests to both endpoints vs.
the previous behavior of waiting, and forwarding the request.
The default is disabled keeping with the old behavior.
Jaco Kroon [Wed, 4 Dec 2019 08:35:52 +0000 (10:35 +0200)]
ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging.
Due to use in res_rtp_asterisk there is a need to be able to apply an
ACL without logging any invalid/denies. It's probably sensible to at
least validate the ACL once directly after load and report invalid ACLs.
add missing argument "rtt" and "status" to the documentation
The change to the dtd file allow an enumlist to contain one or many
configOptionToEnum or enum.
This is different from the previous patch I submitted when you could have a
configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or
(one or many enums)
George Joseph [Wed, 4 Dec 2019 21:01:22 +0000 (14:01 -0700)]
res_rtp_asterisk: Add frame list cleanups to ast_rtp_read
In Asterisk 16+, there are a few places in ast_rtp_read where we've
allocated a frame list but return a null frame instead of the list.
In these cases, any frames left in the list won't be freed. In the
vast majority of the cases, the list is empty when we return so
there's nothing to free but there have been leaks reported in the
wild that can be traced back to frames left in the list before
returning.
The escape paths now all have logic to free frames left in the
list.
Jaco Kroon [Wed, 4 Dec 2019 14:35:35 +0000 (16:35 +0200)]
chan_sip: in case of tcp/tls, be less annoying about tx errors.
chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to
213.150.203.60:1492 returned -2: Interrupted system call
returned -2 implies this wasn't actually an OS error, so errno makes no
sense either. Internal error was already logged higher up, and -2
generally means that either there isn't a valid connection available, or
the pipe notification failed, and that is already correctly logged.
George Joseph [Mon, 26 Aug 2019 02:20:13 +0000 (20:20 -0600)]
res_pjsip_nat: Restore original contact for REGISTER responses
RFC3261 Section 10 "Registrations", specifically paragraph
"10.2.4: Refreshing Bindings", states that a user agent compares
each contact address (in a 200 REGISTER response) to see if it
created the contact. If the Asterisk endpoint has the
rewrite_contact option set however, the contact host and port sent
back in the 200 response will be the rewritten one and not the
one sent by the user agent. This prevents the user agent from
matching its own contact. Some user agents get very upset when
this happens and will not consider the registration successful.
While this is rare, it is acceptable behavior especially if more
than 1 user agent is allowed to register to a single endpoint/aor.
This commit updates res_pjsip_nat (where rewrite_contact is
implemented) to store the original incoming Contact header in
a new "x-ast-orig-host" URI parameter before rewriting it, and to
restore the original host and port to the Contact headers in the
outgoing response.
This is only done if the request is a REGISTER and rewrite_contact
is enabled.
pjsip_message_filter was also updated to ensure that if a request
comes in with any existing x-ast-* URI parameters, we remove them
so they don't conflict. Asterisk will never send a request
with those headers in it but someone might just decide to add them
to a request they craft and send to Asterisk.
NOTE: If a device changes its contact address and registers again,
it's a NEW registration. If the device didn't unregister the
original registration then all existing behavior based
on aor/remove_existing and aor/max_contacts apply.
ASTERISK-28502 Reported-by: Ross Beer
Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e
Sean Bright [Wed, 4 Dec 2019 21:26:46 +0000 (16:26 -0500)]
res_pjsip_registrar.c: Prevent potential double free if AOR is not found
The simple fix here is simply to NULL out username and password after we call
ast_free on them. Unfortunately, I noticed that we weren't checking for
allocation failures for username and password, and adding those checks made
things noisy and cumbersome.
So instead we partially rollback the recent LGTM patch, and move the alloca
calls into find_aor_name().
Sean Bright [Wed, 4 Dec 2019 21:12:39 +0000 (16:12 -0500)]
res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases
We're appropriately sizing the id_domain_alias buffer, but then copying the data
into the id_domain one. We were then using the uninitialized id_domain_alias
buffer we just allocated.
This is ASTERISK~28641 adjacent, but significant enough to warrant its own
patch.
Frederic LE FOLL [Wed, 27 Nov 2019 18:11:33 +0000 (19:11 +0100)]
chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.
During capabilities selection (joint capabilities of us and peer,
configured capability for this peer, or general configured
capabilities), if sip_new() does not keep framing information,
then directmedia activation will fail for any framing different
from default framing.
George Joseph [Fri, 22 Nov 2019 16:39:36 +0000 (09:39 -0700)]
channel.c: Resolve issue with receiving SIP INFO packets for DTMF
The problem is essentially the same as in ASTERISK~28245. Besides
the direct media scenario we have an additional scenario where a
special client is involved. This device mutes audio by default in
transmit direction (no rtp frames) and activates audio only by a
foot switch. In this situation dtmf input (pin for conferences,
transfer features codes , etc) using SIP INFO mode is not
understood properly especially when SIP INFO messages are sent
quickly.
This patch ensures that SIP INFO frames are properly queued and
processed in the above scenario. The patch also corrects situations
where successive dtmf events are received quicker than the
signalled event duration (plus minimum gap/pause) allows, i.e. DTMF
events have to be buffered in the ast channel read queue and
emulation has to be processed asynchronously at slower speed.
Reported by: Thomas Arimont
patches:
trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525)
Salah Ahmed [Fri, 15 Nov 2019 17:34:26 +0000 (11:34 -0600)]
res_pjsip_t38: T.38 error correction mode selection at 200 ok received
if asterisk offer T38 SDP with none error correction scheme and
the endpoint respond with redundancy EC scheme, asterisk switch
to that mode. Since we configure the endpoint as none EC mode
we should not switch to any other mode except none.
following logic implemented in code.
1. If asterisk offer none, and anything except none in answer
will be ignored.
2. If asterisk offer fec, answer with fec, redundancy and none will
be accepted.
3. If asterisk offer redundancy, answer with redundancy and none
will be accepted.
Ben Ford [Mon, 21 Oct 2019 19:55:06 +0000 (14:55 -0500)]
chan_sip.c: Prevent address change on unauthenticated SIP request.
If the name of a peer is known and a SIP request is sent using that
peer's name, the address of the peer will change even if the request
fails the authentication challenge. This means that an endpoint can
be altered and even rendered unusuable, even if it was in a working
state previously. This can only occur when the nat option is set to the
default, or auto_force_rport.
This change checks the result of authentication first to ensure it is
successful before setting the address and the nat option.
George Joseph [Thu, 24 Oct 2019 17:41:23 +0000 (11:41 -0600)]
manager.c: Prevent the Originate action from running the Originate app
If an AMI user without the "system" authorization calls the
Originate AMI command with the Originate application,
the second Originate could run the "System" command.
projects / thirdparty / asterisk.git / log
