Allow mapping pjproject log messages to the Asterisk TRACE
log level. The defaults were also changes to log pjproject
levels 3,4 to DEBUG and 5,6 to TRACE. Previously 3,4,5,6
all went to DEBUG.
ASTERISK-29582
Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d
Commit 305ce3d added -Wno-parentheses-equality to Makefile.rules,
turning the previous two warning suppressions from commit e9520db
redundant. Let us remove the latter.
Change-Id: I0b471254b31e6e05902062761dded4b3e626c7ac
app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
cdr_mysql was deprecated in 1.8, to be removed in 19.
app_mysql was deprecated in 1.8, to be removed in 19.
app_ices was deprecated in 16, to be removed in 19.
app_macro was deprecated in 16, to be removed in 21.
app_fax was deprecated in 16, to be removed in 19.
app_url was deprecated in 16, to be removed in 19.
app_image was deprecated in 16, to be removed in 19.
app_nbscat was deprecated in 16, to be removed in 19.
app_dahdiras was deprecated in 16, to be removed in 19.
cdr_syslog was deprecated in 16, to be removed in 19.
chan_oss was deprecated in 16, to be removed in 19.
chan_phone was deprecated in 16, to be removed in 19.
chan_sip was deprecated in 17, to be removed in 21.
chan_nbs was deprecated in 16, to be removed in 19.
chan_misdn was deprecated in 16, to be removed in 19.
chan_vpb was deprecated in 16, to be removed in 19.
res_config_sqlite was deprecated in 16, to be removed in 19.
res_monitor was deprecated in 16, to be removed in 21.
conf2ael was deprecated in 16, to be removed in 19.
muted was deprecated in 16, to be removed in 19.
ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29554
ASTERISK-29555
ASTERISK-29557
ASTERISK-29558
ASTERISK-29559
ASTERISK-29560
ASTERISK-29561
ASTERISK-29562
ASTERISK-29563
ASTERISK-29564
ASTERISK-29565
ASTERISK-29566
ASTERISK-29567
ASTERISK-29568
ASTERISK-29569
ASTERISK-29570
ASTERISK-29571
ASTERISK-29572
ASTERISK-29573
ASTERISK-29574
Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131
PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request.
It may be used to get all X- headers in case the actual set and names of headers unknown.
ASTERISK-29389
Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b
Meter types are not well supported,
lacking support in telegraf, datadog and the official statsd servers.
We deprecate meters and provide a compliant fallback for any existing usages.
A flag has been introduced to allow meters to fallback to counters.
ASTERISK-29513
Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7
If a re-INVITE is received after we have sent a BYE request then it
is possible for no channel to be present on the session. If this
occurs we allow PJSIP to produce the offer instead. Since the call
is being hung up if it produces an incorrect offer it doesn't
actually matter. This also ensures that code which produces SDP
does not need to handle if a channel is not present.
ASTERISK-29381
Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042
Verify `ast_check_hangup` before looping to the next sound file.
If the call is already hangup we just break the cycle.
It also ensures that the PlaybackFinished event is sent if the call was hangup.
This is also use-full when we are playing a big list of file for a channel that is hangup.
Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played.
With the patch we just break the playback cycle when the chan is hangup.
ASTERISK-29501 #close
Change-Id: Ic4e1c01b974c9a1f2d9678c9d6b380bcfc69feb8
From RFC 8225 Section 5.2.1:
The "dest" claim is a JSON object with the claim name of "dest"
and MUST have at least one identity claim object. The "dest"
claim value is an array containing one or more identity claim JSON
objects representing the destination identities of any type
(currently "tn" or "uri"). If the "dest" claim value array
contains both "tn" and "uri" claim names, the JSON object should
list the "tn" array first and the "uri" array second. Within the
"tn" and "uri" arrays, the identity strings should be put in
lexicographical order, including the scheme-specific portion of
the URI characters.
Additionally, make it clear that there was a failure to sign the JWT
payload and not necessarily a memory allocation failure.
Change-Id: Ia8733b861aef6edfaa9c2136e97b447a01578dc9
Use the URI parsing functions to parse playback URLs in order to find
their file extensions.
For backwards compatibility, we first look at the full URL, then at
any Content-Type header, and finally at just the path portion of the
URL.
ASTERISK-27871 #close
Change-Id: I16d0682f6d794be96539261b3e48f237909139cb
Add check that data parameter specified when audiosocket used for externalMedia.
ASTERISK-29514 #close
Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617
In f8b0c2c9 we added support for port numbers in 'match' statements
but neglected to include that support in the PJSIP config wizard.
The removed code would have also prevented IPv6 addresses from being
successfully used in the config wizard as well.
ASTERISK-29503 #close
Change-Id: Idd5bbfd48009e7a741757743dbaea68e2835a34d
When we try to play a list of sound files in the same Play command,
we get only one PlaybackFinish event, after all sounds are played.
But in the case where the Play fails (because channel is destroyed
for example), Asterisk will send one PlaybackFinish event for each
sound file still to be played. If the list is big, Asterisk is
sending many events.
This patch adds a failed state so we can understand that the play
failed. On that case we don't send the event, if we still have a
list of sounds to be played.
When we reach the last sound, we send the PlaybackFinish with
the failed state.
ASTERISK-29464 #close
Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322
With the fix for ASTERISK_28754 channels are no longer put on hold if an
outbound INVITE is answered with a "Session Progress" containing
"inactive" audio.
The previous change moved the evaluation of the media attributes to
`negotiate_incoming_sdp_stream()` to have the `remotely_held` status
available when building the SDP in `create_outgoing_sdp_stream()`.
This however means that an answer to an outbound INVITE, which does not
traverse `negotiate_incoming_sdp_stream()`, cannot set the
`remotely_held` status anymore.
This change moves the check so that both, `negotiate_incoming_sdp_stream()` and
`apply_negotiated_sdp_stream()` can do the checks.
ASTERISK-29479
Change-Id: Icde805a819399d5123b688e1ed1d2bcd9d5b0f75
When the MessageSend destination is in the form
PJSIP/<number>@<endpoint> and the endpoint's contact
URI already has a user component, that user component
will now be replaced with <number> when creating the
request URI.
ASTERISK_29404
Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5
Set preferred transport when querying the local address to use in
filter_on_tx_messages(). This prevents the module to erroneously select
the wrong transport if more than one transports of the same type (TCP or
TLS) are configured.
ASTERISK-29241
Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6
The text description needs to be the last thing on the AST_MODULE_INFO
line to be pulled in properly by menuselect.
Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832
* Implemented the new "to" parameter of the MessageSend()
dialplan application. This allows a user to specify
a complete SIP "To" header separate from the Request URI.
* Completely refactored the get_outbound_endpoint() function
to actually handle all the destination combinations that
we advertized as supporting.
* We now also accept a destination in the same format
as Dial()... PJSIP/number@endpoint
* Added lots of debugging.
ASTERISK-29404
Reported by Brian J. Murrell
Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce
STIR/SHAKEN requires a Date header alongside the Identity header, so
that has been added. Still on the outgoing side, we were missing the
dest->tn section of the JSON payload, so that has been added as well.
Moving to the incoming side, URL checking has been added to the public
cert URL to ensure that it starts with http.
https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c
For connection oriented transports PJSIP uses factories to
produce transports. When doing a partial transport reload
we need to also move the factory of the transport over so
that anything referencing the transport (such as an endpoint)
has the factory available.
ASTERISK-29441
Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161
When unsubscribing from an endpoint technology a FRACK
would occur due to incorrect reference counting. This fixes
that issue, along with some other issues.
Fixed a typo in get_subscription when calling ao2_find as it
needed to pass the endpoint ID and not the entire object.
Fixed scenario where a subscription would get returned when
it shouldn't have been when searching based on endpoint
technology.
A doulbe unreference has also been resolved by only explicitly
releasing the reference held by tech_subscriptions.
ASTERISK-28237 #close
Reported by: Lucas Tardioli Silveira
Change-Id: Ia91b15f8e5ea68f850c66889a6325d9575901729
In multidomain environments, it is desirable to create
PJSIP endpoints with the domain info in the endpoint name
in pjsip_endpoint.conf. This resulted in an error with
registrations, NOTIFY, and OPTIONS packet generation.
This commit will detect if there is an @ in the endpoint
identifier and generate the URI accordingly so NOTIFY and
OPTIONS From headers will generate correctly.
ASTERISK-28393
Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619
RTCP ICE candidates use a base address derived from the RTP
candidate. The port on the base address was not being updated to
the RTCP port.
This change sets the base port to the RTCP port and all is well.
ASTERISK-29433
Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040
By default Asterisk reports the PJSIP version in a SOFTWARE attribute
of every STUN packet it sends. This may not be desired in a production
environment, and RFC5389 recommends making the use of the SOFTWARE
attribute a configurable option:
https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2
This patch adds a `stun_software_attribute` yes/no option to make it
possible to omit the SOFTWARE attribute from STUN packets.
ASTERISK-29434
Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b
RFC7616 and RFC8760 allow more than one WWW-Authenticate or
Proxy-Authenticate header per realm, each with different digest
algorithms (including new ones like SHA-256 and SHA-512-256).
Thankfully however a UAS can NOT send back multiple Authenticate
headers for the same realm with the same digest algorithm. The
UAS is also supposed to send the headers in order of preference
with the first one being the most preferred. We're supposed to
send an Authorization header for the first one we encounter for a
realm that we can support.
The UAS can also send multiple realms, especially when it's a
proxy that has forked the request in which case the proxy will
aggregate all of the Authenticate headers and then send them all
back to the UAC.
It doesn't stop there though... Each realm can require a
different username from the others. There's also nothing
preventing each digest algorithm from having a unique password
although I'm not sure if that adds any benefit.
So now... For each Authenticate header we encounter, we have to
determine if we support the digest algorithm and, if not, just
skip the header. We then have to find an auth object that
matches the realm AND the digest algorithm or find a wildcard
object that matches the digest algorithm. If we find one, we add
it to the results vector and read the next Authenticate header.
If the next header is for the same realm AND we already added an
auth object for that realm, we skip the header. Otherwise we
repeat the process for the next header.
In the end, we'll have accumulated a list of credentials we can
pass to pjproject that it can use to add Authentication headers
to a request.
NOTE: Neither we nor pjproject can currently handle digest
algorithms other than MD5. We don't even have a place for it in
the ast_sip_auth object. For this reason, we just skip processing
any Authenticate header that's not MD5. When we support the
others, we'll move the check into the loop that searches the
objects.
Changes:
* Added a new API ast_sip_retrieve_auths_vector() that takes in
a vector of auth ids (usually supplied on a call to
ast_sip_create_request_with_auth()) and populates another
vector with the actual objects.
* Refactored res_pjsip_outbound_authenticator_digest to handle
multiple Authenticate headers and set the stage for handling
additional digest algorithms.
* Added a pjproject patch that allows them to ignore digest
algorithms they don't support. This patch has already been
merged upstream.
* Updated documentation for auth objects in the XML and
in pjsip.conf.sample.
* Although res_pjsip_authenticator_digest isn't affected
by this change, some debugging and a testsuite AMI event
was added to facilitate testing.
Discovered during OpenSIPit 2021.
ASTERISK-29397
Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281
RFC 4235 Section 4.1.6 describes XML elements that should be
sent to subscribed endpoints to identify the local and remote
participants in the dialog.
This patch adds this functionality to PJSIP by iterating through the
ringing channels causing the NOTIFY, and inserts the channel info
into the dialog so that information is properly passed to the endpoint
in dialog-info+xml.
ASTERISK-24601
Patch submitted: Joshua Elson
Modified by: Joseph Nadiv and Sean Bright
Tested by: Joseph Nadiv
Change-Id: I20c5cf5b45f34d7179df6573c5abf863eb72964b
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.
https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c
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.
https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
Change-Id: Ia6e1a4028c1933a0e1d204b769ebb9f5a11f00ab
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.
https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d
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.
Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a
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.
ASTERISK-29030 #close
Change-Id: Ie70d657d580d9a1f2877e25a6ef161c5ad761cf7
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.
ASTERISK-29130
Reported-By: Francisco Correia
Reported-By: BJ Weschke
Reported-By: Sébastien Duthil
Change-Id: If0c8e40853bc0e9429f2ba9c7f5f358d90c311af
For some input to the standard deviation algorithm extremely large,
and wrong numbers were being calculated.
This patch uses a new formula for correctly calculating both the
running mean and standard deviation for the given inputs.
ASTERISK-29364 #close
Change-Id: Ibc6e18be41c28bed3fde06d612607acc3fbd621f
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.
Change-Id: I07d6e21cec61e289c2326138d6bcbcb3c3d5e008
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.
ASTERISK-29373
Change-Id: I1accd9c6e859811336e670e698bd8bd124f33226
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.
Change-Id: I25fa7ae9bad6ed68e273bbcccf0ee51cae6e7024
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.
ASTERISK-29215
Change-Id: I61313cca7fc571144338aac826091791b87b6e17
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.
ASTERISK-29354
Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a
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.
ASTERISK-29352
Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a
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.
ASTERISK-29336
Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35
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.
ASTERISK-29335
Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90
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.
Change-Id: Idf2a73457be231647aed3c87b1da197afba86892
