Naveen Albert [Thu, 13 May 2021 14:47:08 +0000 (10:47 -0400)]
app_voicemail: Configurable voicemail beep
Hitherto, VoiceMail() played a non-customizable beep tone to indicate
the caller could leave a message. In some cases, the beep may not
be desired, or a different tone may be desired.
To increase flexibility, a new option allows customization of the tone.
If the t option is specified, the default beep will be overridden.
Supplying an argument will cause it to use the specified file for the tone,
and omitting it will cause it to skip the beep altogether. If the option
is not used, the default behavior persists.
Naveen Albert [Thu, 13 May 2021 15:32:06 +0000 (11:32 -0400)]
AMI: Add AMI event to expose hook flash events
Although Asterisk can receive and propogate flash events, it currently
provides no mechanism for doing anything with them itself.
This AMI event allows flash events to be processed by Asterisk.
Additionally, AST_CONTROL_FLASH is included in a switch statement
in channel.c to avoid throwing a warning when we shouldn't.
Naveen Albert [Thu, 13 May 2021 15:13:55 +0000 (11:13 -0400)]
main/file.c: Don't throw error on flash event.
AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c
where it should be ignored. Adding this to the switch ensures a
warning isn't thrown on RFC2833 flash events, since nothing's amiss.
Ben Ford [Mon, 26 Apr 2021 22:00:11 +0000 (17:00 -0500)]
STIR/SHAKEN: Switch to base64 URL encoding.
STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.
The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.
Ben Ford [Tue, 11 May 2021 17:26:13 +0000 (12:26 -0500)]
STIR/SHAKEN: OPENSSL_free serial hex from openssl.
We're getting the serial number of the certificate from openssl and
freeing it with ast_free(), but it needs to be freed with OPENSSL_free()
instead. Now we duplicate the string and free the one from openssl with
OPENSSL_free(), which means we can still use ast_free() on the returned
string.
Ben Ford [Wed, 21 Apr 2021 16:12:55 +0000 (11:12 -0500)]
STIR/SHAKEN: Fix certificate type and storage.
During OpenSIPit, we found out that the public certificates must be of
type X.509. When reading in public keys, we use the corresponding X.509
functions now.
We also discovered that we needed a better naming scheme for the
certificates since certificates with the same name would cause issues
(overwriting certs, etc.). Now when we download a public certificate, we
get the serial number from it and use that as the name of the cached
certificate.
The configuration option public_key_url in stir_shaken.conf has also
been renamed to public_cert_url, which better describes what the option
is for.
George Joseph [Thu, 22 Apr 2021 18:07:22 +0000 (12:07 -0600)]
Updates for the MessageSend Dialplan App
Enhancements:
* The MessageSend dialplan application now takes an optional
third argument that can set the message's "To" field on
outgoing messages. It's an alternative to using the
MESSAGE(to) dialplan function.
NOTE: No channel driver currently implements this field. A
follow-on commit for res_pjsip_messaging will implement it for
the chan_pjsip channel driver.
* To prevent confusion with the first argument, currently named
"to", it's been renamed to "destination". Its function,
creating the request URI, hasn't changed.
* The documentation for MessageSend was updated to be
more clear about the parameters and how they interact
the MESSAGE() dialplan function.
* With the rename of MessageSend's first parameter, and the fact
that message.c references <info> elements in chan_sip.c,
res_pjsip_messaging.c and res_xmpp, they each needed
documentation updates to use MessageDestinationInfo instead of
MessageToInfo.
* appdocsxml.dtd was updated to include a missing element
declaration for "dataType". This was showing up as an error
in Eclipse's dtd editor.
* Despite the changes in this commit, there should be
no impact to current users of MessageSend.
chan_local: Skip filtering audio formats on removed streams.
When a stream topology is provided to chan_local when dialing
it filters the audio formats down. This operation did not skip
streams which were removed (that have no formats) resulting in
calling being aborted.
Sean Bright [Tue, 27 Apr 2021 17:31:30 +0000 (13:31 -0400)]
res_rtp_asterisk: More robust timestamp checking
We assume that a timestamp value of 0 represents an 'uninitialized'
timestamp, but 0 is a valid value. Add a simple wrapper to be able to
differentiate between whether the value is set or not.
This also removes the fix for ASTERISK~28812 which should not be
needed if we are checking the last timestamp appropriately.
George Joseph [Fri, 2 Apr 2021 12:21:33 +0000 (06:21 -0600)]
bridge_channel_write_frame: Check for NULL channel
There is a possibility, when bridge_channel_write_frame() is
called, that the bridge_channel->chan will be NULL. The first
thing bridge_channel_write_frame() does though is call
ast_channel_is_multistream() which had no check for a NULL
channel and therefore caused a segfault. Since it's still
possible for bridge_channel_write_frame() to write the frame to
the other channels in the bridge, we don't want to bail before we
call ast_channel_is_multistream() but we can just skip the
multi-channel stuff. So...
bridge_channel_write_frame() only calls ast_channel_is_multistream()
if bridge_channel->chan is not NULL.
As a safety measure, ast_channel_is_multistream() now returns
false if the supplied channel is NULL.
ASTERISK-29379 Reported-by: Vyrva Igor Reported-by: Ross Beer
Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce
George Joseph [Thu, 1 Apr 2021 13:39:03 +0000 (07:39 -0600)]
res_prometheus: Clone containers before iterating
The channels, bridges and endpoints scrape functions were
grabbing their respective global containers, getting the
count of entries, allocating metric arrays based on
that count, then iterating over the container. If the
global container had new objects added after the count
was taken and the metric arrays were allocated, we'd run
out of metric entries and attempt to write past the end
of the arrays.
Now each of the scape functions clone their respective
global containers and all operations are done on the
clone. Since the clone is stable between getting the
count and iterating over it, we can't run past the end
of the metrics array.
Joshua C. Colp [Wed, 10 Mar 2021 15:03:11 +0000 (11:03 -0400)]
loader: Output warnings for deprecated modules.
Using the information from the MODULEINFO XML we can
now output useful information at the end of module
loading for deprecated modules. This includes the
version it was deprecated in, the version it will be
removed in, and the replacement if available.
Kevin Harwell [Mon, 29 Mar 2021 22:40:49 +0000 (17:40 -0500)]
res_rtp_asterisk: Don't count 0 as a minimum lost packets
The calculated minimum lost packets represents the lowest number of
lost packets missed during an RTCP report interval. Zero of course
is the lowest, but the idea is that this value contain the lowest
number of lost packets once some have been missed.
This patch checks to make sure the number of lost packets over an
interval is not zero before checking and setting the minimum value.
Also, this patch updates the rtp lost packet test to check for
packet loss over several reports vs one.
Joshua C. Colp [Mon, 29 Mar 2021 22:52:08 +0000 (19:52 -0300)]
res_rtp_asterisk: Only raise flash control frame on end.
Flash in RTP is conveyed the same as DTMF, just with a
specific digit. In Asterisk however we do flash as a
single control frame.
This change makes it so that only on end do we provide
the flash control frame to the core. Previously we would
provide a flash control frame on both begin and end,
causing flash to work improperly.
Kevin Harwell [Fri, 5 Mar 2021 18:53:47 +0000 (12:53 -0600)]
res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command
This patch makes it so when Asterisk is compiled in DEVMODE a CLI
command is available that allows someone to drop incoming RTP
packets. The command allows for dropping of packets once, or on a
timed interval (e.g. drop 10 packets every 5 seconds). A user can
also specify to drop packets by IP address.
Kevin Harwell [Fri, 5 Mar 2021 18:47:38 +0000 (12:47 -0600)]
time: Add timeval create and unit conversion functions
Added a TIME_UNIT enumeration, and a function that converts a
string to one of the enumerated values. Also, added functions
that create and initialize a timeval object using a specified
value, and unit type.
Sean Bright [Tue, 23 Mar 2021 20:15:45 +0000 (16:15 -0400)]
app_queue.c: Remove dead 'updatecdr' code.
Also removed the sample documentation, and some oddly-placed
documentation about the timeout argument to the Queue() application
itself. There is a large section on the timeout behavior below.
Mark Murawski [Fri, 19 Mar 2021 14:11:26 +0000 (10:11 -0400)]
logger: Console sessions will now respect logger.conf dateformat= option
The 'core' console (ie: asterisk -c) does read logger.conf and does
use the dateformat= option.
Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
and uses a hard coded dateformat option for printing received verbose messages:
main/logger.c: static char dateformat[256] = "%b %e %T"
This change will load logger.conf for each remote console session and
use the dateformat= option to set the per-line timestamp for verbose messages
Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1
ASTERISK-25358: #close Reported-by: Igor Liferenko
George Joseph [Thu, 18 Mar 2021 16:14:34 +0000 (10:14 -0600)]
res_pjsip_session: Make reschedule_reinvite check for NULL topologies
When the check for equal topologies was added to reschedule_reinvite()
it was assumed that both the pending and active media states would
actually have non-NULL topologies. We since discovered this isn't
the case.
We now only test for equal topologies if both media states have
non-NULL topologies. The logic had to be rearranged a bit to make
sure that we cloned the media states if their topologies were
non-NULL but weren't equal.
Joshua C. Colp [Fri, 19 Mar 2021 09:56:18 +0000 (06:56 -0300)]
app_queue: Only send QueueMemberStatus if status changes.
If a queue member was updated with the same status multiple
times each time a QueueMemberStatus event would be sent
which would be a duplicate of the previous.
This change makes it so that the QueueMemberStatus event is
only sent if the status actually changes.
Joshua C. Colp [Mon, 1 Mar 2021 23:32:24 +0000 (19:32 -0400)]
res_pjsip: Add support for partial transport reload.
Some configuration items for a transport do not result in
the underlying transport changing, but instead are just
state we keep ourselves and use. It is perfectly reasonable
to change these items.
These include local_net and external_* information.
Joshua C. Colp [Wed, 17 Mar 2021 15:28:45 +0000 (12:28 -0300)]
res_rtp_asterisk: Force resync on SSRC change.
When an SSRC change occurs the timestamps are likely
to change as well. As a result we need to reset the
timestamp mapping done in the calc_rxstamp function
so that they map properly from timestamp to real
time.
This previously occurred but due to packet
retransmission support the explicit setting
of the marker bit was not effective.
Joshua C. Colp [Wed, 10 Mar 2021 14:05:58 +0000 (10:05 -0400)]
menuselect: Add ability to set deprecated and removed versions.
The "deprecated_in" and "removed_in" information can now be
set in MODULEINFO for a module and is then displayed in
menuselect so users can be aware of when a module is slated
to be deprecated and then removed.
Joshua C. Colp [Wed, 10 Mar 2021 10:47:26 +0000 (06:47 -0400)]
documentation: Fix non-matching module support levels.
Some modules have a different support level documented in their
MODULEINFO XML and Asterisk module definition. This change
brings the two in sync for the modules which were not matching.
Joshua C. Colp [Tue, 9 Mar 2021 14:54:27 +0000 (10:54 -0400)]
xml: Embed module information into core XML documentation.
This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.
Alexander Traud [Fri, 5 Mar 2021 17:16:56 +0000 (18:16 +0100)]
res_format_attr_*: Parameter Names are Case-Insensitive.
see RFC 4855:
parameter names are case-insensitive both in media type strings and
in the default mapping to the SDP a=fmtp attribute.
This change is required for H.263+ because some implementations are
known to use even mixed-case. This does not fix ASTERISK~29268 because
H.264 was not fixed. This approach here lowers/uppers both parameter
names and parameter values. H.264 needs a different approach because
one of its parameter values is not case-insensitive:
sprop-parameter-sets is Base64.
Sean Bright [Mon, 8 Mar 2021 23:16:14 +0000 (18:16 -0500)]
res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse.
ao2_replace() bumps the reference count of the object that is doing the
replacing, which is not what we want. We just want to drop the old ref
on the old object and update the pointer to point to the new object.
Torrey Searle [Fri, 19 Feb 2021 11:50:21 +0000 (12:50 +0100)]
res/res_rtp_asterisk: generate new SSRC on native bridge end
For RTCP to work, we update the ssrc to be the one corresponding to
the native bridge while active. However when the bridge ends we
should generate a new SSRC as the sequence numbers will not continue
from the native bridge left off.
Joshua C. Colp [Mon, 1 Mar 2021 21:35:20 +0000 (17:35 -0400)]
sorcery: Add support for more intelligent reloading.
Some sorcery objects actually contain dynamic content
that can change despite the underlying configuration
itself not changing. A good example of this is the
res_pjsip_endpoint_identifier_ip module which allows
specifying hostnames. While the configuration may not
change between reloads the DNS information of the
hostnames can.
This change adds the ability for a sorcery object to be
marked as having dynamic contents which is then taken
into account when reloading by the sorcery file based
config module. If there is an object with dynamic content
then a reload will be forced while if there are none
then the existing behavior of not reloading occurs.
George Joseph [Tue, 2 Mar 2021 18:55:38 +0000 (11:55 -0700)]
res_pjsip_refer: Move the progress dlg release to a serializer
Although the dlg session count was incremented in a pjsip servant
thread, there's no guarantee that the last thread to unref this
progress object was one. Before we decrement, we need to make
sure that this is either a servant thread or that we push the
decrement to a serializer that is one.
Because pjsip_dlg_dec_session requires the dialog lock, we don't
want to wait on the task to complete if we had to push it to a
serializer.
Joshua C. Colp [Wed, 3 Mar 2021 18:31:07 +0000 (14:31 -0400)]
res_pjsip_registrar: Include source IP and port in log messages.
When registering it can be useful to see the source IP address and
port in cases where multiple devices are using the same endpoint
or when anonymous is in use.
Ben Ford [Thu, 25 Feb 2021 19:50:47 +0000 (13:50 -0600)]
AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite.
When Asterisk sends a reinvite negotiating T38 faxing, it's possible a
crash can occur if the response contains a m=image and zero port. The
reinvite callback code now checks session_media to see if it is null or
not before trying to access the udptl variable on it.
Alexander Traud [Thu, 28 Jan 2021 14:39:01 +0000 (15:39 +0100)]
res_format_attr_h263: Generate valid SDP fmtp for H.263+.
Fixed:
* RFC 4629 does not allow the value "0" for MPI, K, and N.
* Allow value "0" for PAR.
* BPP is printed only when specified because "0" has a meaning.
New:
* Added CPCF and MaxBR.
* Some implementations provide CIF without MPI: a=fmtp:xx CIF;F=1
Although a violation of RFC 3555 section 3, we can support that.
Changed:
* Resorts the CIFs from large to small which partly fixes ASTERISK~29267.
Joshua C. Colp [Wed, 24 Feb 2021 13:04:09 +0000 (09:04 -0400)]
res_pjsip_nat: Don't rewrite Contact on REGISTER responses.
When sending a SIP response to an incoming REGISTER request
we don't want to change the Contact header as it will
contain the Contacts registered to the AOR and not our own
Contact URI.
Joshua C. Colp [Wed, 3 Mar 2021 13:32:22 +0000 (09:32 -0400)]
channel: Fix memory leak in suppress API.
A frame suppression API exists as part of channels
which allows audio frames to or from a channel to
be dropped. The MuteAudio AMI action uses this
API to perform its job.
This API uses a framehook to intercept flowing
audio and drop it when appropriate. It is the
responsibility of the framehook to free the
frame it is given if it changes the frame. The
suppression API failed to do this resulting in
a leak of audio frames.
Salah Ahmed [Wed, 27 Jan 2021 20:01:01 +0000 (14:01 -0600)]
res_rtp_asterisk: Check remote ICE reset and reset local ice attrb
This change will check is the remote ICE session got reset or not by
checking the offered ufrag and password with session. If the remote ICE
reset session then Asterisk reset its local ufrag and password to reject
binding request with Old ufrag and Password.
Nico Kooijman [Sun, 28 Feb 2021 09:24:29 +0000 (10:24 +0100)]
main: With Dutch language year after 2020 is not spoken in say.c
Implemented the english way of saying the year in ast_say_date_with_format_nl.
Currently the numbers are spoken correctly until 2020 and stopped working
this year.
ASTERISK-29297 #close Reported-by: Jacek Konieczny
Change-Id: If5918eed5ab05df31df4dd23f08a909a60f6aba4
Nick French [Thu, 25 Feb 2021 02:51:55 +0000 (20:51 -0600)]
res_pjsip: dont return early from registration if init auth fails
If set_outbound_initial_authentication_credentials() fails,
handle_client_registration() bails early without creating or
sending a register message.
[set_outbound_initial_authentication_credentials() failures
can occur during the process of retrieving an oauth access
token.]
The return from handle_client_registration is ignored, so
returning an error doesn't do any good.
This is a real problem when the registration request is a
re-register, because then the registration will still be
marked 'active' despite the re-register never being sent at all.
So instead, log a warning but let the registration be created
and sent (and probably fail) and follow the normal registration
failed retry/abort logic.
George Joseph [Fri, 19 Feb 2021 19:25:13 +0000 (12:25 -0700)]
res_pjsip_refer: Refactor progress locking and serialization
Although refer_progress_notify() always runs in the progress
serializer, the pjproject evsub module itself can cause the
subscription to be destroyed which then triggers
refer_progress_on_evsub_state() to clean it up. In this case,
it's possible that refer_progress_notify() could get the
subscription pulled out from under it while it's trying to use
it.
At one point we tried to have refer_progress_on_evsub_state()
push the cleanup to the serializer and wait for its return before
returning to pjproject but since pjproject calls its state
callbacks with the dialog locked, this required us to unlock the
dialog while waiting for the serialized cleanup, then lock it
again before returning to pjproject. There were also still some
cases where other callers of refer_progress_notify() weren't
using the serializer and crashes were resulting.
Although all callers of refer_progress_notify() now use the
progress serializer, we decided to simplify the locking so we
didn't have to unlock and relock the dialog in
refer_progress_on_evsub_state().
Now, refer_progress_notify() holds the dialog lock for its
duration and since pjproject also holds the dialog lock while
calling refer_progress_on_evsub_state() (which does the cleanup),
there should be no more chances for the subscription to be
cleaned up while still being used to send NOTIFYs.
To be extra safe, we also now increment the session count on
the dialog when we create a progress object and decrement
the count when the progress is destroyed.
Ben Ford [Mon, 15 Feb 2021 18:24:42 +0000 (12:24 -0600)]
res_pjsip_session.c: Check topology on re-invite.
Removes an unnecessary check for the conditional that compares the
stream topologies to see if they are equal to suppress re-invites. This
was a problem when a Digium phone received an INVITE that offered codecs
different than what it supported, causing Asterisk to send the
re-invite.
Jaco Kroon [Tue, 23 Feb 2021 11:28:08 +0000 (13:28 +0200)]
res_odbc_transaction: correctly initialise forcecommit value from DSN.
Also improve the in-process documentation to clarify that the value is
initialised from the DSN and not default false, but that the DSN's value
is default false if unset.
Joshua C. Colp [Tue, 16 Feb 2021 18:33:07 +0000 (14:33 -0400)]
res_pjsip_session: Always produce offer on re-INVITE without SDP.
When PJSIP receives a re-INVITE without an SDP offer the INVITE
session library will first call the on_create_offer callback and
if unavailable then use the active negotiated SDP as the offer.
In some cases this would result in a different SDP then was
previously used without an incremented SDP version number. The two
known cases are:
1. Sending an initial INVITE with a set of codecs and having the
remote side answer with a subset. The active negotiated SDP would
have the pruned list but would not have an incremented SDP version
number.
2. Using re-INVITE for unhold. We would modify the active negotiated
SDP but would not increment the SDP version.
To solve these, and potential other unknown cases, the on_create_offer
callback has now been implemented which produces a fresh offer with
incremented SDP version number. This better fits within the model
provided by the INVITE session library.
Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.
This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.
Jaco Kroon [Wed, 17 Feb 2021 20:51:17 +0000 (22:51 +0200)]
func_odbc: Introduce minargs config and expose ARGC in addition to ARGn.
minargs enables enforcing of minimum count of arguments to pass to
func_odbc, so if you're unconditionally using ARG1 through ARG4 then
this should be set to 4. func_odbc will generate an error in this case,
so for example
[FOO]
minargs = 4
and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
potentially leaked ARG4 from Gosub().
ARGC is needed if you're using optional argument, to verify whether or
not an argument has been passed, else it's possible to use a leaked ARGn
from Gosub (app_stack). So now you can safely do
${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
George Joseph [Tue, 9 Feb 2021 17:25:27 +0000 (10:25 -0700)]
res_pjsip_refer: Always serialize calls to refer_progress_notify
refer_progress_notify wasn't always being called from the progress
serializer. This could allow clearing notification->progress->sub
in one thread while another was trying to use it.
* Instances where refer_progress_notify was being called in-line,
have been changed to use ast_sip_push_task().
Kevin Harwell [Mon, 1 Feb 2021 21:24:25 +0000 (15:24 -0600)]
AST-2021-002: Remote crash possible when negotiating T.38
When an endpoint requests to re-negotiate for fax and the incoming
re-invite is received prior to Asterisk sending out the 200 OK for
the initial invite the re-invite gets delayed. When Asterisk does
finally send the re-inivite the SDP includes streams for both audio
and T.38.
This happens because when the pending topology and active topologies
differ (pending stream is not in the active) in the delayed scenario
the pending stream is appended to the active topology. However, in
the fax case the pending stream should replace the active.
This patch makes it so when a delay occurs during fax negotiation,
to or from, the audio stream is replaced by the T.38 stream, or vice
versa instead of being appended.
Further when Asterisk sent the re-invite with both audio and T.38,
and the endpoint responded with a declined T.38 stream then Asterisk
would crash when attempting to change the T.38 state.
This patch also puts in a check that ensures the media state has a
valid fax session (associated udptl object) before changing the
T.38 state internally.
Ivan Poddubnyi [Mon, 28 Dec 2020 12:43:23 +0000 (13:43 +0100)]
res_pjsip_diversion: Fix adding more than one histinfo to Supported
New responses sent within a PJSIP sessions are based on those that were
sent before. Therefore, adding/modifying a header once causes it to be
sent on all responses that follow.
Sending 181 Call Is Being Forwarded many times first adds "histinfo"
duplicated more and more, and eventually overflows past the array
boundary.
This commit adds a check preventing adding "histinfo" more than once,
and skipping it if there is no more space in the header.
Similar overflow situations can also occur in res_pjsip_path and
res_pjsip_outbound_registration so those were also modified to
check the bounds and suppress duplicate Supported values.
Joshua C. Colp [Fri, 5 Feb 2021 11:26:02 +0000 (07:26 -0400)]
pjsip: Make modify_local_offer2 tolerate previous failed SDP.
If a remote side is broken and sends an SDP that can not be
negotiated the call will be torn down but there is a window
where a second 183 Session Progress or 200 OK that is forked
can be received that also attempts to negotiate SDP. Since
the code marked the SDP negotiation as being done and complete
prior to this it assumes that there is an active local and remote
SDP which it can modify, while in fact there is not as the SDP
did not successfully negotiate. Since there is no local or remote
SDP a crash occurs.
This patch changes the pjmedia_sdp_neg_modify_local_offer2
function to no longer assume that a previous SDP negotiation
was successful.
Ben Ford [Mon, 11 Jan 2021 20:20:34 +0000 (14:20 -0600)]
core_unreal: Fix T.38 faxing when using local channels.
After some changes to streams and topologies, receiving fax through
local channels stopped working. This change adds a stream topology with
a stream of type IMAGE to the local channel pair and allows fax to be
received.
Alexander Traud [Fri, 5 Feb 2021 08:33:41 +0000 (09:33 +0100)]
chan_sip: Allow [peer] without audio (text+video).
Two previous commits, 620d9f4 and 6d980de, allow to set up a call
without audio, again. That was introduced originally with commit f04d5fb
but changed and broke over time. The original commit missed one
scenario: A [peer] section in sip.conf, which does not allow audio at
all. In that case, chan_sip rejected the call, although even when the
requester offered no audio. Now, chan_sip does not check whether there
is no audio format but checks whether there is no format in general. In
other words, if there is at least one format to offer, the call succeeds.
However, to prevent calls with no-audio, chan_sip still rejects calls
when both call parties (caller = requester of the call *and* callee =
[peer] section in sip.conf) included audio. In such a case, it is
expected that the call should have audio.
George Joseph [Thu, 28 Jan 2021 18:02:34 +0000 (11:02 -0700)]
chan_iax2.c: Require secret and auth method if encryption is enabled
If there's no secret specified for an iax2 peer and there's no secret
specified in the dial string, Asterisk will crash if the auth method
requested by the peer is MD5 or plaintext. You also couldn't specify
a default auth method in the [general] section of iax.conf so if you
don't have static peers defined and just use the dial string, Asterisk
will still crash even if you have a secret specified in the dial string.
* Added logic to iax2_call() and authenticate_reply() to print
a warning and hanhup the call if encryption is requested and
there's no secret or auth method. This prevents the crash.
* Added the ability to specify a default "auth" in the [general]
section of iax.conf.
Alexander Traud [Wed, 27 Jan 2021 17:42:06 +0000 (18:42 +0100)]
chan_sip: Set up calls without audio (text+video), again.
The previous commit 6d980de fixed this issue in the core of Asterisk.
With that, each channel technology can be used without audio
theoretically. Practically, the channel-technology driver chan_sip
turned out to have an invalid check preventing that. chan_sip tested
whether there is at least one audio format. However, chan_sip has to
test whether there is at least one format. More cannot be tested while
requesting chan_sip because only the [general] capabilities but not the
[peer] caps are known yet. And the [peer] caps might not be a subset or
show any intersection with the [general] caps. This change here fixes
this.
The original commit f04d5fb, thirteen years ago, contained a software
bug as it passed ANY audio capability to the channel-technology driver.
Instead, it should have passed NO audio format. Therefore, this
addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8.
Then, Asterisk 10 changed that from ANY to NO, but nobody reported since
then.
When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received
This allows applications to perform actions based on the failure
reason.
ASTERISK-29252 #close Reported-by: Dan Cropp
Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4
roadkill [Fri, 22 Jan 2021 13:38:01 +0000 (14:38 +0100)]
res/res_pjsip.c: allow user=phone when number contain *#
if From number contain * or # asterisk will not add user=phone
Currently only number that uses AST_DIGIT_ANYNUM can have "user=phone" but the validation should use AST_DIGIT_ANY
this is a problem when you want to send call to ISUP
as they will disregard the From header and either replace From with anonymous or with p-asserted-identity
ASTERISK-29261
Reported by: Mark Petersen
Tested by: Mark Petersen
Alexander Traud [Thu, 21 Jan 2021 19:28:06 +0000 (20:28 +0100)]
chan_sip: SDP: Reject audio streams correctly.
This completes the fix for ASTERISK_24543. Only when the call is an
outgoing call, consult and append the configured format capabilities
(p->caps). When all audio formats got rejected the negotiated format
capabilities (p->jointcaps) contain no audio formats for incoming
calls. This is required when there are other accepted media streams.
projects / thirdparty / asterisk.git / log
