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
This changes the outgoing offer call preference
default option to match the behavior of previous
versions of Asterisk.
The additional advanced codec negotiation options
have also been removed from the sample configuration
and marked as reserved for future functionality in
XML documentation.
The codec preference options have also been fixed to
enforce local codec configuration.
ASTERISK-29109
Change-Id: Iad19347bd5f3d89900c15ecddfebf5e20950a1c2
When handling a send_message request to a non-existing endpoint, the response's
body is overriden and not properly freed.
ASTERISK-29108
Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8
Added debug logging categories that allow a user to output debug
information based on a specified category. This lets the user limit,
and filter debug output to data relevant to a particular context,
or topic. For instance the following categories are now available for
debug logging purposes:
dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet,
stun, stun_packet
These debug categories can be enable/disable via an Asterisk CLI command.
While this overrides, and outputs debug data, core system debugging is
not affected by this patch. Statements still output at their appropriate
debug level. As well backwards compatibility has been maintained with
past debug groups that could be enabled using the CLI (e.g. rtpdebug,
stundebug, etc.).
ASTERISK-29054 #close
Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849
In the event that the desired extension already exists,
ast_add_extension2_lockopt() will free the 'data' it is passed before
returning an error, so we should not be freeing it ourselves.
Additionally, there were two places where ast_add_extension2_lockopt()
could return an error without also freeing the 'data' pointer, so we
add that.
ASTERISK-29097 #close
Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae
Stop advertising RFC2833 support on the rtp_engine when DTMF mode is
auto but no tel_event was found inside SDP file.
On an incoming call create_rtp will be called and when session->dtmf is
set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without
looking at the SDP file.
Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND
but continued to advertise RFC2833 support.
This meant the native_rtp bridge would falsely consider the two channels
as compatible. In addition to changing the DTMF mode we now set or
remove the AST_RTP_PROPERTY_DTMF.
The property is checked in ast_rtp_dtmf_compatible and called by
native_rtp_bridge_compatible.
ASTERISK-29051 #close
Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287
Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and
AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice,
resulting in to 181 being generated.
Change-Id: I866e5461564644ffb8a5e12b6f1330b50a7b63ab
When constructing a stream name based on the media type
and position the allocated name was not being freed
causing a leak.
Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de
Only track our sample offset if we are playing a non-announcement file,
otherwise we will skip that number of samples when we start playing the
first MoH file.
ASTERISK-24329 #close
Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc
The ast_sip_dialog_get_session function returns the session
with reference count increased. This was not taken into
account and was causing sessions to remain around when they
should not be.
ASTERISK-29089
Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8
Sometimes not play MOH on bridge.
ASTERISK-29081
Reported-by: Michal Hajek <michal.hajek@daktela.com>
Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232
The module description needs to be on the same line as the
AST_MODULE_INFO or it is not parsed correctly.
Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21
Implemention of History-Info capable of interworking with Diversion
Header following RFC7544
ASTERISK-29027 #close
Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6
The recent 491 changes introduced a check to determine if the active
and pending topologies were equal and to suppress the re-invite if they
were. When a re-invite is sent for a COLP-only change, the pending
topology is NULL so that check doesn't happen and the re-invite is
correctly sent. Of course, sending the re-invite sets the pending
topology. If a 491 is received, when we resend the re-invite, the
pending topology is set and since we didn't request a change to the
topology in the first place, pending and active topologies are equal so
the topologies-equal check causes the re-invite to be erroneously
suppressed.
This change checks if the topologies are equal before we run the media
state resolver (which recreates the pending topology) so that when we
do the final topologies-equal check we know if this was a topology
change request. If it wasn't a change request, we don't suppress
the re-invite even though the topologies are equal.
ASTERISK-29014
Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314
Added to:
* bridges/bridge_softmix.c
* channels/chan_pjsip.c
* include/asterisk/res_pjsip_session.h
* main/channel.c
* res/res_pjsip_session.c
There NO functional changes in this commit.
Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3
When both Asterisk and a UA send re-invites at the same time, both
send 491 "Transaction in progress" responses to each other and back
off a specified amount of time before retrying. When Asterisk
prepares to send its re-invite, it sets up the session's pending
media state with the new topology it wants, then sends the
re-invite. Unfortunately, when it received the re-invite from the
UA, it partially processed the media in the re-invite and reset
the pending media state before sending the 491 losing the state it
set in its own re-invite.
Asterisk also was not tracking re-invites received while an existing
re-invite was queued resulting in sending stale SDP with missing
or duplicated streams, or no re-invite at all because we erroneously
determined that a re-invite wasn't needed.
There was also an issue in bridge_softmix where we were using a stream
from the wrong topology to determine if a stream was added. This also
caused us to erroneously determine that a re-invite wasn't needed.
Regardless of how the delayed re-invite was triggered, we need to
reconcile the topology that was active at the time the delayed
request was queued, the pending topology of the queued request,
and the topology currently active on the session. To do this we
need a topology resolver AND we need to make stream named unique
so we can accurately tell what a stream has been added or removed
and if we can re-use a slot in the topology.
Summary of changes:
* bridge_softmix:
* We no longer reset the stream name to "removed" in
remove_all_original_streams(). That was causing multiple streams
to have the same name and wrecked the checks for duplicate streams.
* softmix_bridge_stream_sources_update() was checking the old_stream
to see if it had the softmix prefix and not considering the stream
as "new" if it did. If the stream in that slot has something in it
because another re-invite happened, then that slot in old might
have a softmix stream but the same stream in new might actually
be a new one. Now we check the new_stream's name instead of
the old_stream's.
* stream:
* Instead of using plain media type name ("audio", "video", etc) as
the default stream name, we now append the stream position to it
to make it unique. We need to do this so we can distinguish multiple
streams of the same type from each other.
* When we set a stream's state to REMOVED, we no longer reset its
name to "removed" or destroy its metadata. Again, we need to
do this so we can distinguish multiple streams of the same
type from each other.
* res_pjsip_session:
* Added resolve_refresh_media_states() that takes in 3 media states
and creates an up-to-date pending media state that includes the changes
that might have happened while a delayed session refresh was in the
delayed queue.
* Added is_media_state_valid() that checks the consistency of
a media state and returns a true/false value. A valid state has:
* The same number of stream entries as media session entries.
Some media session entries can be NULL however.
* No duplicate streams.
* A valid stream for each non-NULL media session.
* A stream that matches each media session's stream_num
and media type.
* Updated handle_incoming_sdp() to set the stream name to include the
stream position number in the name to make it unique.
* Updated the ast_sip_session_delayed_request structure to include both
the pending and active media states and updated the associated delay
functions to process them.
* Updated sip_session_refresh() to accept both the pending and active
media states that were in effect when the request was originally queued
and to pass them on should the request need to be delayed again.
* Updated sip_session_refresh() to call resolve_refresh_media_states()
and substitute its results for the pending state passed in.
* Updated sip_session_refresh() with additional debugging.
* Updated session_reinvite_on_rx_request() to simply return PJ_FALSE
to pjproject if a transaction is in progress. This stops us from
creating a partial pending media state that would be invalid later on.
* Updated reschedule_reinvite() to clone both the current pending and
active media states and pass them to delay_request() so the resolver
can tell what the original intention of the re-invite was.
* Added a large unit test for the resolver.
ASTERISK-29014
Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb
Currently, it was not possible to create bridge with video_mode single.
This made hard to put the bridge in a vidoe_single mode.
So, added video_single option for Bridge creation using the ARI.
This allows create a bridge with video_mode single.
ASTERISK-29055
Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae
Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again.
Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6
When fixing issues uncovered by GCC10 a copy of the parker UUID
was removed accidentally. This change restores it so that the
subscription has the data it needs.
ASTERISK-29042
Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a
Properly bump reference on format object to avoid memory corruption on double free
ASTERISK-29040 #close
Change-Id: Ic5a7faabfe2ef965ddb024186e1de7ca4542e2a3
Adapt the response handler so it also called when 181 is received.
In the case 181 is received, also generate the 181 response.
ASTERISK-29001 #close
Change-Id: I73cfee46a8ca85371280ebdb38674f8fde7510df
Per the RFC when an outgoing re-INVITE is done we should
only terminate the dialog if a 481 or 408 is received.
ASTERISK-29033
Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503
Two changes of note in this patch:
* Use ast_file_read_dir instead of opendir/readdir/closedir
* If the files list should be sorted, do that at the end rather than as
we go which improves performance for large lists
Change-Id: Ic7e9c913c0f85754c99c74c9cf6dd3514b1b941f
The MoH class internal file vector is potentially being manipulated by
multiple threads at the same time without sufficient locking. Switch to
a reference counted list and operate on copies where necessary.
ASTERISK-28927 #close
Change-Id: I479c5dcf88db670956e8cac177b5826c986b0217
When reading in a codec preference configuration option
the value would be set on the respective option before
applying any default adjustments, resulting in the
configuration not being as expected.
This was exposed by the REST API push configuration as
it used the configuration returned by Asterisk to then do
a modification. In the case of codec preferences one of
the options had a transcode value of "unspecified" when the
defaults should have ensured it would be "allow" instead.
This also renames the options in other places that were
missed.
Change-Id: I4ad42e74fdf181be2e17bc75901c62591d403964
This change renames the codec preference endpoint options.
incoming_offer_codec_prefs becomes codec_prefs_incoming_offer
to keep the options together when showing an endpoint.
Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d
Sean Bright
George Joseph
Ben Ford
Kevin Harwell
Joshua C. Colp
Jaco Kroon
Alexander Traud
Torrey Searle
Salah Ahmed
Nick French
Alexei Gradinari
Boris P. Korzun
Ivan Poddubnyi
roadkill
Robert Cripps
Jean Aunis
Richard Mudgett
Sungtae Kim
Pirmin Walthert
Nathan Bruning
Stanislav
Alexander Greiner-Baer
Kevin Harwell
Holger Hans Peter Freyther
lvl
Michal Hajek
Patrick Verzele
Nickolay Shmyrev