Fixes the following compile error:
chan_vpb.cc:2688:26: error: catching polymorphic type
‘class std::exception’ by value
Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649
Given a scenario where MixMonitor was initiated over AMI it
was possible for the channel and MixMonitor thread to remain
alive past hang up of the channel. This scenario required
the AMI initiated MixMonitor to retrieve the channel, a
hangup to occur on the channel in another thread, and then
for MixMonitor to actually start. If this occurred the
MixMonitor thread would remain alive indefinitely and
the channel reference would remain.
This change ensures that audiohooks are never able to
be attached to channels that have been hung up. An
additional fix has also been done in app_mixmonitor to
properly release the channel reference if this occurs.
ASTERISK-28780
Change-Id: I8044c06daa06f0f16607788c596f55623be26f58
This is a generic jenkinsfile to build Asterisk and optionally
perform one or more of the following:
* Publish the API docs to the wiki
* Run the Unit tests
* Run Testsuite Tests
This job can be triggered manually from Jenkins or be triggered
automatically on a schedule based on a cron string.
Change-Id: Id9d22a778a1916b666e0e700af2b9f1bacda0852
bridge_p2p_rtp_write will forward rtp to the bridged rtp instance
without modifying the ssrc. However, it is not updating the SSRC
in the bridged rtp. Thus, when SSRC packets are generated, they
have the correct SSRC for the sender.
ASTERISK-28773 #close
Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478
If ICE support is enabled but not negotiated, the rtp->ice structure is
not being destroyed. This leads to Asterisk waiting for ICE to complete
instead of immediately starting the DTLS handshake, resulting in the
call leg having no RTP.
ASTERISK-28769 #close
Change-Id: I17c137546dc9ecfb9583c24dcf4c2ced8bbd7a27
If the SSRC of a received RTP packet differed from the previous SSRC
an SSRC change control frame would be queued ahead of the media
frame. In the case of audio this would result in the format of the
audio frame not being checked, and if it differed or was not allowed
then it could cause the call to drop due to failure to set up a
translation path.
The chan_pjsip module will now no longer assume the first frame
will be the audio frame and instead goes through the complete list
to find it.
ASTERISK-28759
Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec
A regular expression in a NAPTR response record can have a trailing
'i' flag to indicate that the expression should be evaluated in a
case-insensitive way. We were not checking for that flag which caused
the record parsing to fail on otherwise valid input.
Although this change will initially go into Asterisk 13, 16, and 17,
it is my intention to replace the majority of this code in 16 and up -
including this fix - by changing enum.c to consume the new DNS API
which duplicates most of this logic already. Asterisk 13 doesn't have
the DNS API, so this fix will be as good as it gets.
ASTERISK-26711 #close
Reported by: Vitold
Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0
This change introduce a CLI command for the RTP to display the general
configuration.
In the first step add the follow fields of the configurations:
- rtpstart
- rtpend
- dtmftimeout
- rtpchecksum
- strictrtp
- learning_min_sequential
- icesupport
Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f
The ast_get_txt() API function (and by extension, the TXTCIDNAME
dialplan function) were broken in
65b8381550 such that we would never
actually make a DNS TXT query as described.
This patch restores the documented behavior.
ASTERISK-19460 #close
Reported by: George Joseph
Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376
When building check_expr2 with ASLR PIE hardening enabled the linker
fails. This is resolved by adding the regular compiler flags when
building the object files from ast_expr2f.c and ast_expr2.c.
Note: The STANDALONE define is removed because it is already defined in
_ASTCFLAGS. YY_NO_INPUT is defined so that the compile survives
'--enable-dev-mode'.
Also, a Makefile variable "CROSS_COMPILING" is added so that the
build system doesn't try to run check_expr2 when cross-compiling,
because that will fail the build as will.
ASTERISK-28685 #close
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
Change-Id: If435b7db9f9ad8266245bda51c81c220f9658915
The receive buffer will now grow if we end up flushing the
receive queue after not receiving the expected packet in time.
This is done in hopes that if this is encountered again the
extra buffer size will allow more time to pass and any missing
packets to be received.
The send buffer will now grow if we are asked for packets and
can't find them. This is done in hopes that the packets are
from the past and have simply been expired. If so then in
the future with the extra buffer space the packets should be
available.
Sequence number cycling has been handled so that the
correct sequence number is calculated and used in
various places, including for sorting packets and
for determining if a packet is old or not.
NACK sending is now more aggressive. If a substantial number
of missing sequence numbers are added a NACK will be sent
immediately. Afterwards once the receive buffer reaches 25%
a single NACK is sent. If the buffer continues to grow and
reaches 50% or greater a NACK will be sent for each received
future packet to aggressively ask the remote endpoint to
retransmit.
ASTERISK-28764
Change-Id: I97633dfa8a09a7889cef815b2be369f3f0314b41
When a text message was received any associated variable was not written to
the ARI TextMessageReceived event. This occurred because Asterisk only wrote
out "send" variables. However, even those "send" variables would fail ARI
validation due to a TextMessageVariable formatting bug.
Since it seems the TextMessageReceived event has never been able to include
actual variables it was decided to remove the TextMessageVariable object type
from ARI, and simply return a JSON object of key/value pairs for variables.
This aligns more with how the ARI sendMessage handles variables, and other
places in ARI.
That being the case, and since this is technically an API breaking change (no
one should really be affected since things never really worked) the ARI version
was updated to reflect that.
ASTERISK-28755 #close
Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f
Update the state of remote_hold immediately on receipt of remote
SDP so that the information is available when building the SDP
answer
ASTERISK-28754 #close
Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f
There are exceptions for plural objects, but they are detected using the
supplied NUMBER, not using an extra option.
Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113
There were a couple places where the format cap function parameter was not
'const' when it should have been. This patch makes them 'const'.
Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e
Warnings without this:
res_config_mysql.c: In function 'update2_mysql':
res_config_mysql.c:741:15: warning: format '%llu' expects argument of type
'long long unsigned int', but argument 6 has type 'my_ulonglong'
{aka 'long unsigned int'} [-Wformat=]
ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n",
numrows, tablename);
(reformatted for readability within line-wrap)
Change-Id: I2af4d419a37c1a7eeee750cf9ae4a9a2b3a37fd3
This change extends the Sorcery API to allow a wizard to be
told to explicitly reload objects or a specific object type
even if the wizard believes that nothing has changed.
This has been leveraged by res_pjsip and res_pjsip_acl to
reload endpoints and PJSIP ACLs when a named ACL changes.
ASTERISK-28697
Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b
Dump OpenSSL's error stack to the error log when things fail.
ASTERISK-28750 #close
Reported by: Martin Zeh
Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a
In order to retry outbound registrations for some situations, we
need access to the tdata from the original request. For instance,
for 401/407 responses we need it to properly construct the
subsequent request with the authentication. We also need it if
we're iterating over a DNS SRV response record set so we can skip
entries we've already tried.
We've been getting the tdata from the server response rdata and
transaction but that only works for the failures where there was
actually a response (4XX, 5XX, etc). For timeouts there's no
response and therefore no rdata or transaction from which to get
the tdata. When processing a single A/AAAA record for a server,
this wasn't an issue as we just retried that same server after the
retry timer expired. If we got an SRV record set for the server
though, without the state from the tdata, we just kept trying the
first entry in the set repeatedly instead of skipping to the next
one in the list.
* Added a "last_tdata" member to the client state structure to keep
track of the sent tdata.
* Updated registration_client_send() to save the tdata it used into
the client_state.
* Updated sip_outbound_registration_response_cb() to use the tdata
saved in client_state when we don't get a response from the
server. We still use the tdata from the transaction when we DO
get a response from the server so we can properly handle 4XX
responses where our new request depends on it.
General note on timeouts:
Although res_pjsip_outbound_registration skips to the next record
immediately when a timeout occurs during SRV set traversal, it's
pjproject that determines how long to wait before a timeout is
declared. As with other SIP message types, pjproject will continue
trying the same server at an interval specified by "timer_t1" until
"timer_b" expires. Both of those timers are set in the pjsip.conf
"system" section.
ASTERISK-28746
Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
This change adds support to bridge_softmix to allow the addition
and removal of additional video source streams. When such a change
occurs each participant is renegotiated as needed to reflect the
update. If another video source is added then each participant
gets another source. If a video source is removed then it is
removed from each participant. This functionality allows you to
have both your webcam and screenshare providing video if you
desire, or even more streams. Mapping has been changed to use
the topology index on the source channel as a unique identifier
for outgoing participant streams, this will never change and
provides an easy way to establish the mapping.
The bridge_simple and bridge_native_rtp modules have also been
updated to renegotiate when the stream topology of a party changes
allowing the same behavior to occur as added to bridge_softmix.
If a screen share is added then the opposite party is renegotiated.
If that screen share is removed then the opposite party is
renegotiated again.
Some additional fixes are also included in here. Stream state is
now conveyed in SDP so sendonly/recvonly/inactive streams can
be requested. Removed streams now also remove previous state
from themselves so consumers don't get confused.
ASTERISK-28733
Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5
When handling ICE negotiations, it's possible that there can be a delay
between STUN binding requests which in turn will cause a delay in ICE
completion, preventing media from flowing. It should be possible to send
media when there is at least one valid pair, preventing this scenario
from occurring.
A change was added to PJPROJECT that adds an optional callback
(on_valid_pair) that will be called when the first valid pair is found
during ICE negotiation. Asterisk uses this to start the DTLS handshake,
allowing media to flow. It will only be called once, either on the first
valid pair, or when ICE negotiation is complete.
ASTERISK-28716
Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867
