Previously there was no way to specify a connection timeout when
attempting to connect a websocket client to a server. This patch
makes it possible to now do such.
Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d
The code currently checks to see if an RFC3389
warning flag is set, except if it is, it merely
sets the flag again, the logic of which doesn't
make any sense.
This adjusts the if comparison to check if the
flag has NOT been set, and if so, emit a notice
log event and set the flag so that future frames
do not cause an event to be logged.
ASTERISK-29856 #close
Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1
Neither pjsip_message_filter's filter_on_tx_message() nor
res_pjsip_session's session_outgoing_nat_hook() were multipart
aware and just assumed that an SDP would be the only thing in
a message body. Both were changed to use the new
pjsip_get_sdp_info() function which searches for an sdp in
both single- and multi- part message bodies.
ASTERISK-29813
Change-Id: I8f5b8cfdc27f1d4bd3e7491ea9090951a4525c56
Added two new functions to assist checking media types...
* ast_sip_are_media_types_equal compares two pjsip_media_types.
* ast_sip_is_media_type_in tests if one media type is in a list
of others.
Added static definitions for commonly used media types to
res_pjsip.h.
Changed several modules to use the new functions and static
definitions.
ASTERISK_29813
(not ready to close)
Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9
There are times when you need to troubleshoot issues with bundled
pjproject or add new features that need to be pushed upstream
but...
* The source directory created by extracting the pjproject tarball
is not scanned for code changes so you have to keep forcing
rebuilds.
* The source directory isn't a git repo so you can't easily create
patches, do git bisects, etc.
* Accidentally doing a make distclean will ruin your day by wiping
out the source directory, and your changes.
* etc.
This commit makes that easier.
See third-party/pjproject/README-hacking.md for the details.
ASTERISK-29824
Change-Id: Idb1251040affdab31d27cd272dda68676da9b268
The ast_rtp_codecs_payloads functions do not preserve the order in which
the payloads were specified on an incoming SDP media line. This leads to
a problem with the codec negotiation functionality, as the format
capabilities of the stream are extracted from the ast_rtp_codecs. This
commit moves the ast_rtp_codec to ast_format conversion to the place
where the order is still known.
ASTERISK-28863
ASTERISK-29320
Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25
res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined
that rtpstart was configured to be an odd value. Also adding a loop
counter to prevent a possible infinite loop when looking for a free
port.
ASTERISK-27406
Change-Id: I90f07deef0716da4a30206e9f849458b2dbe346b
Newer versions of spandsp did refactoring of code to add new features
like color FAXing. This refactoring broke backwards compatibility.
Add support for the new version while retaining support for 0.0.6.
ASTERISK-29729 #close
Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1
Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.
This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.
ASTERISK-29777 #close
Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b
Makes basic call progress tone detection available
in a tech-agnostic manner with the addition of the
ToneScan application. This can determine if the channel
has encountered a busy signal, SIT tones, dial tone,
modem, fax machine, etc. A few basic async progress
tone detect options are also added to the TONE_DETECT
function.
ASTERISK-29720 #close
Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90
Fixes four misuses of the parameter 'name'. Additionally, for
consistency and to avoid such an issue in future, those few other
places, which used '\file name', were changed just to '\file'. Then,
Doxygen uses the name of the current file.
ASTERISK-29733
Change-Id: I0c18b4c863c6988b138c77448057349a9ee7052d
Fix parsing of ANI2/OLI information, since it was previously
parsing the user, when it should have been parsing other_param.
Also improves the parsing by using pjproject native functions
rather than trying to parse the parameters ourselves like
chan_sip did. A previous attempt at this caused a crash, but
this works correctly now.
ASTERISK-29703 #close
Change-Id: I8f3c79032d9ea1a21d16f8e11f22bd8d887738a1
The stir_shaken configuration option now has 4 different choices to pick
from: off, attest, verify, and on. Off and on behave the same way they
do now. Attest will only perform attestation on the endpoint, and verify
will only perform verification on the endpoint.
Certain responses are required to be sent based on certain conditions
for STIR/SHAKEN. For example, if we get a Date header that is outside of
the time range that is considered valid, a 403 Stale Date response
should be sent. This and several other responses have been added.
Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7
Add a new function that converts a speech results type to a string.
Also add another function to unregister an engine, but returns a
pointer to the unregistered engine object instead of a success/fail
integer.
Change-Id: I0f7de17cb411021c09fb03988bc2b904e1380192
In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
fallback use of the transport's bind address solve problems sending
media on systems that cannot send ipv4 packets on ipv6 sockets, and
certain other situations. This change extends both of these behaviors
to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
problems on these systems, introducing a new option
endpoint/t38_bind_udptl_to_media_address.
ASTERISK-29402
Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557
The behavior of max_contacts and remove_existing are connected. If
remove_existing is enabled, the soonest expiring contacts are removed.
This may occur when there is an unavailable contact. Similarly,
when remove_existing is not enabled, registrations from good
endpoints are rejected in favor of retaining unavailable contacts.
This commit adds a new AOR option remove_unavailable, and the effect
of this setting will depend on remove_existing. If remove_existing
is set to no, we will still remove unavailable contacts when they
exceed max_contacts, if there are any. If remove_existing is set to
yes, we will prioritize the removal of unavailable contacts before
those that are expiring soonest.
ASTERISK-29525
Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784
When listing bridges we go through the ones present in
ARI, get their snapshot, turn it into JSON, and add it
to the payload we ultimately return.
An invisible "dial bridge" exists within ARI that would
also try to be added to this payload if the channel
"create" and "dial" routes were used. This would ultimately
fail due to invisible bridges having no snapshot
resulting in the listing of bridges failing.
This change makes it so that the listing of bridges
ignores invisible ones.
ASTERISK-29668
Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a
Rather than stripping parameters from Content-Type headers before
comparison, first try to compare the whole string. If no match is
found, strip the parameters and try that way.
ASTERISK-29275 #close
Change-Id: I2963c8ecbb3a9605b78b6421c415108d77a66a0f
Some code has been added referencing symbols defined in a block
protected by #ifdef HAVE_PJPROJECT. Protect those code parts in
ifdef blocks too.
ASTERISK-29660
Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f
Adds parsing of ANI II digits (Originating
Line Information) to PJSIP, on par with
what currently exists in chan_sip.
ASTERISK-29472
Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847
dsp.c contains arbitrary tone detection functionality
which is currently only used for fax tone recognition.
This change makes this functionality publicly
accessible so that other modules can take advantage
of this.
Additionally, a WaitForTone and TONE_DETECT app and
function are included to allow users to do their
own tone detection operations in the dialplan.
ASTERISK-29546
Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26
With gcc 11, res/res_snmp.c and res/snmp/agent.c need the
-fPIC option added to its _ASTCFLAGS.
ASTERISK-29634
Change-Id: I34649c85e075fd954e578378fabf798c3f038f50
When compiled without extended srtp crypto suites also disable parsing
these from received SDP. This prevents using these, as some client
implementations are not stable.
ASTERISK-29625
Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a
This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.
The refresh of the name resolution occurs first when the module is
loaded, then recurringly, slightly after the previous DNS answer TTL
expires.
ASTERISK-29508 #close
Change-Id: I7955a046293f913ba121bbd82153b04439e3465f
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.
app_macro was deprecated in 16, to be removed in 21.
chan_sip was deprecated in 17, to be removed in 21.
res_monitor was deprecated in 16, to be removed in 21.
ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29558
ASTERISK-29567
ASTERISK-29572
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
Asterisk first looks at the end of the URL to determine the file
extension of the returned audio, which in many cases will not work
because the URL may end with a query string or a URL fragment. If that
fails, Asterisk then looks at the Content-Type header and then finally
parses the URL to get the extension.
The order has been changed such that we look at the Content-Type
header first, followed by looking for the extension of the parsed
URL. We no longer look at the end of the URL, which was error prone.
ASTERISK-29527 #close
Change-Id: I1e3f83b339ef2b80661704717c23568536511032
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 cURL's URL parsing API, falling back to the urlparser library, 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
Added support for a basic AEAP configuration read from aeap.conf.
Also added 2 CLI commands for showing individual configurations as
well as all of them: aeap show server <id> and aeap show servers.
Only one configuration option is required at the moment, and that one is
server_url. It must be a websocket URL. The other option, codecs, is
optional and will be used over the codecs specified on the endpoint if
provided.
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=45482453
Change-Id: I567ac5148c92b98d29d2ad83421b416b75ffdaa3
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
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
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
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
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.
Pointed out by George Joseph in #asterisk-dev
Change-Id: Ie8167ed3d4b52b9d1ea2d785f885e8c27206743d
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.
ASTERISK-29300 #close
Change-Id: I23334b6934d2bf6490bda4bbf6414d96b8d17d10
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.
ASTERISK-29321
Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1
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.
Change-Id: I8ff2d5d94be3ff04298394070434e22a7d3cbc41
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.
ASTERISK-29325
Change-Id: Ie178a6f55f53f8473035854c411bc3d056e0a2e0
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.
ASTERISK-29305
Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9
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.
Change-Id: I95a650c715007b8dde11a77cb37d9c6c123a441e
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.
ASTERISK-29235
Change-Id: I35a0723545281dd01fcd5cae497baab58720478c
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.
ASTERISK-29266
Change-Id: I9c55e79a7af98a8fbb497d336b828ba41bc34eeb
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.
ASTERISK-29315 #close
Change-Id: I2e03b1ea7fba1fa1a8279086aa4b17679e7fa7fa
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.
ASTERISK-29313
Change-Id: I97a8bb01771a3c85345649b8124507f7622a8480
For some RTCP packet types the report count is actually the packet's subtype.
This was not being reflected in the packet debug output.
This patch makes it so for some RTCP packet types a "Packet Subtype" is
now output in the debug replacing the "Reception reports" (i.e count).
Change-Id: Id4f4b77bb37077a4c4f039abd6a069287bfefcb8
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.
ASTERISK-28452
Change-Id: I2d81048d54edcb80fe38fdbb954a86f0a58281a1
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.
ASTERISK-29311 #close
Change-Id: I46e2379f7b0656034442bce77cb37ccd4e61098d
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
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.
ASTERISK-29303
Change-Id: I04dc91befb2387904e28a9aaeaa3bcdbcaa7fa63
Added a SELECT 'LIMIT' clause to realtime_pgsql() and refactored the function.
ASTERISK-29293 #close
Change-Id: If5a6d4b1072ea2e6e89059b21139d554a74b34f5
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.
ASTERISK-29203 #close
Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09
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.
ASTERISK-29227
Reported by: Ivan Poddubny
Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322
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().
Change-Id: Idcf1934c4e873f2c82e2d106f8d9f040caf9fa1e
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
Change-Id: I3307bdbf757582740bfee4110e85f7b6c9291cc4
Provided a support of variuos URL-schemes for res_musiconhold,
registered by ast_bucket_scheme_register().
ASTERISK-29262 #close
Change-Id: If0ea8697587353dce358a70035d82649fd4632b6
The last argument to ast_copy_string() is the buffer size, not the
number of characters, so we add 1 to avoid stamping out the final \n
in the persisted SUBSCRIBE message.
Change-Id: I019b78942836f57965299af15d173911fcead5b2
function ast_sip_session_media_state_add.
Check ast_media_type matches when a ast_sip_session_media is found
otherwise when transitioning from say image to audio, the wrong
session is returned in the first if statement.
ASTERISK-29220 #close
Change-Id: I6f6efa9b821ebe8881bb4c8c957f8802ddcb4b5d
When both a tech subscription and an endpoint subscription exist for a given
endpoint, TextMessageReceived events are dispatched to the tech subscription
only.
ASTERISK-29229
Change-Id: I9eac4cba5f9e27285a282509395347abc58fc2b8
session->channel doesn't exist until chan_pjsip creates it, so intead of
setting a channel variable every new incoming call sets one and the same
global variable.
This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
a newly created channel, it also removes a misleading reference to
channel->session used to fetch call pickup configuraion.
ASTERISK-29240
Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755
This reverts commit 2fe76dd816.
Reason for revert: Too many issues reported. Need to research and correct.
ASTERISK-29230
ASTERISK-29231
Reported by: Michael Maier
Change-Id: I6453af680e17d8ffe7af2c5de7e1b2a58c8793cb
Segfault occurs during outbound UDP registration when all
transport states are being iterated over. The transport object
in the transport is accessed, but flow transports have a NULL
transport object.
Modify to not iterate over any flow transport
ASTERISK-29210 #close
Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a
Add a check to see if the URI is a Tel URI and prevent crashing on
trying to retrieve the reason parameter.
ASTERISK-29191
ASTERISK-29219
Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37
(cherry picked from commit a7aea71e60)
AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an
unsigned long on all machines.
Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338
The documentation in the wiki says there should be spyee-channel
information elements in the ChanSpyStop AMI event.
https://wiki.asterisk.org/wiki/x/Xc5uAg
However, this is not the case in Asterisk <= 16.10.0 Version. We're
using these Spyee* arguments since Asterisk 11.x, so these arguments
vanished in Asterisk 12 or higher.
For maximum compatibility, we still send the ChanSpyStop event even if
we are not able to find any 'Spyee' information.
ASTERISK-28883 #close
Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f
In rewrite_uri asterisk was not making deep copies of strings when
changing the uri. This was in some cases causing garbage in the route
header and in other cases even crashing asterisk when receiving a
message with a record-route header set. Thanks to Ralf Kubis for
pointing out why this happens. A similar problem was found in
res_pjsip_transport_websocket.c. Pjproject needs as well to be patched
to avoid garbage in CANCEL messages.
ASTERISK-29024 #close
Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b
This adds support for both Digium and Sangoma user agent strings
for the Sangoma specific body supplement.
Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482
In some circumstances it was possible for an INVITE
session to be destroyed while we were still using it.
This occurred due to the reference on the INVITE session
being released internally as a result of its state
changing to DISCONNECTED.
This change adds a reference to the INVITE session
which is released when our own session is destroyed,
ensuring that the INVITE session remains valid for
the lifetime of our session.
ASTERISK-29022
Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9
By default libcurl does not follow redirects, so we explicitly enable
it by setting CURLOPT_FOLLOWLOCATION. Once that is enabled, libcurl
will follow up to CURLOPT_MAXREDIRS redirects, which by default is
configured to be unlimited.
This patch sets CURLOPT_MAXREDIRS to a more reasonable default (8). If
we determine at some point that this needs to be increased on
configurable it is a trivial change.
ASTERISK-29173 #close
Change-Id: I4925ebbcf0c7d728bb9252b3795b3479ae225b30
the 'J' is missing in module description.
"PSIP STIR/SHAKEN Module for Asterisk" -> "PJSIP STIR/SHAKEN Module for Asterisk"
ASTERISK-29175 #close
Change-Id: I17da008540ee2e8496b644d05f995b320b54ad7a
RFC 3261 says that the Accept-Encoding header should be present
in an options response. Permitted values according to RFC 2616
are only compression algorithms like gzip or the default identity
encoding. Therefore "text/plain" is not a correct value here.
As long as the header is hard coded, it should be set to "identity".
Without this fix an Alcatel OmniPCX periodically logs warnings like
"[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed"
on a SIP Trunk.
ASTERISK-29165 #close
Change-Id: I0aa2211ebf0b4c2ed554ac7cda794523803a3840
* Instead of using the pjproject timer heap, we now use our own
pjsip_scheduler. This allows us to more easily debug and allows us to
see times in "pjsip show/list registrations" as well as being able to
see the registrations in "pjsip show scheduled_tasks".
* Added the last registration time, registration interval, and the next
registration time to the CLI output.
* Removed calls to pjsip_regc_info() except where absolutely necessary.
Most of the calls were just to get the server and client URIs for log
messages so we now just save them on the client_state object when we
create it.
* Added log messages where needed and updated most of the existong ones
to include the registration object name at the start of the message.
Change-Id: I4534a0fc78c7cb69f23b7b449dda9748c90daca2
* Added a ONESHOT type that never reschedules.
* Added "like" capability to "pjsip show scheduled_tasks" so you can do
the following:
CLI> pjsip show scheduled_tasks like outreg
PJSIP Scheduled Tasks:
Task Name Interval Times Run ...
============================================= ========= ========= ...
pjsip/outreg/testtrunk-reg-0-00000074 50.000 oneshot ...
pjsip/outreg/voipms-reg-0-00000073 110.000 oneshot ...
* Fixed incorrect display of "Next Start".
* Compacted the displays of times in the CLI.
* Added two new functions (ast_sip_sched_task_get_times2,
ast_sip_sched_task_get_times_by_name2) that retrieve the interval,
next start time, and next run time in addition to the times already
returned by ast_sip_sched_task_get_times().
Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3
This avoids the inclusion of the OpenSSL headers in the public header,
which avoids one external library dependency in res_pjsip_stir_shaken.
Change-Id: I6a07e2d81d2b5442e24e99b8cc733a99f881dcf4
pjproject returns the dialog locked and with a reference. However,
in Asterisk the method that handles this decrements the reference
and removes the lock prior to returning. This makes it possible,
under some circumstances, for another thread to free said dialog
before the thread that created it attempts to use it again. Of
course when the thread that created it tries to use a freed dialog
a crash can occur.
This patch makes it so Asterisk now returns the newly created
dialog both locked, and with an added reference. This allows the
caller to de-reference, and unlock the dialog when it is safe to
do so.
In the case of a new SIP Invite the lock, and reference are now
held for the entirety of the new invite handling process.
Otherwise it's possible for the dialog, or its dependent objects,
like the transaction, to disappear. For example if there is a TCP
transport error.
ASTERISK-29057 #close
Change-Id: I5ef645a47829596f402cf383dc02c629c618969e
(cherry picked from commit 6baa4b53be)
If Asterisk sends out and INVITE and receives a challenge with a
different nonce value each time, it will continually send out INVITEs,
even if the call is hung up. The endpoint must be configured for
outbound authentication in order for this to occur. A limit has been set
on outbound INVITEs so that, once reached, Asterisk will stop sending
INVITEs and the transaction will terminate.
ASTERISK-29013
Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7
This patch initializes a couple of local variables to some default values.
Interestingly, in the 'pj_status_t dlg_status' case the value not being
initialized caused memory to grow, and not be recovered, in the off nominal
path (at least on my machine).
Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7
Commit 44bb0858cb ("debugging: Add enough
to choke a mule") accidentally removed calls to
ast_sip_message_apply_transport when it was attempting to just add
debugging code.
The kiss of death was saying that there were no functional changes in
the commit comment.
This makes outbound calls that use the 'flow' transport mechanism fail,
since this call is used to relay headers into the outbound INVITE
requests.
ASTERISK-29124 #close
Change-Id: I0f3e32c2e8ac415e30b1d29966d75a1546f0526a
