The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to
date.
Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2
GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
ASTERISK-27156 #close
Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
Setting this option will cause the Queue application to only announce
the caller's position if it has improved since the last time that we
announced it.
Change-Id: I173a124121422209485b043e2bf784f54242fce6
OpenSSL has 2 levels or error processing. It's possible for the
top layer to return SSL_ERROR_SYSCALL but the lower layer return
no error, in which case processing should continue. Only the top
layer was being examined though so connections were being torn
down when they didn't need to be. This patch adds the examination
of the lower level codes, and if they return no errors, allows
processing to continue.
ASTERISK-27001
Reported-by: Ian Gilmour
patches:
pjproject-2.6.patch submitted by Ian Gilmour (license 6889)
Updated-by: George Joseph and Sauw Ming (Teluu)
Merged to upstream pjproject on 7/27/2017 (commit 5631)
Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis
ASTERISK-27085 #close
Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
This change does a few things to improve packet loss and renegotiation:
1. On outgoing RTP streams we will now properly reflect out of order
packets and packet loss in the sequence number. This allows the
remote jitterbuffer to better reorder things.
2. Video updates can now be discarded for a period of time
after one has been sent to prevent flooding of clients.
3. For declined and removed streams we will now release any
media session resources associated with them. This was not
previously done and caused an issue where old state was being
used for a new stream.
4. RTP bundling was not actually removing bundled RTP instances
from the parent. This has been resolved by removing based on
the RTP instance itself and not the SSRC.
5. The code did not properly handle explicitly unbundling an
RTP instance from its parent. This now works as expected.
ASTERISK-27143
Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45
Case scenario with Applications ARI:
* Once you subscribe to deviceState with Applications REST API, it will be
added into subscription pool.
* When you unsubscribe it will remove from the device_state_subscription
hash table but not from the subscription pool.
* When you subscribe again, it will add it to pool again.
* Now you will have two subscriptions and you will receive same event
twice.
This fix should now remove deviceState subscription from pool and it
should fix unsubscribe on deviceState.
ASTERISK-27130 #close
Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4
The following testsuite voicemail tests were failing to re-enter the
mailbox after the first login attempt.
tests/apps/voicemail/authenticate_invalid_mailbox
tests/apps/voicemail/authenticate_invalid_password
The tests were noting the start of the vm-incorrect-mailbox prompt and
immediately sending the mailbox for the next login attempt. Since the
invalid message playback had to complete before the digits were
recognized, the test passed for the wrong reason and added approximately
20 seconds to the test times.
* Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox
digits like the initial vm-login prompt so the tests are able to enter the
intended mailbox.
Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8
Incrementing version for the Extra sounds release. 1.5.1 Extra sounds
removes two prompts that were moved into the Core packages in the 1.6 Core
sounds release.
ASTERISK-27142 #close
Change-Id: I82f017812b0ea9599e19dd4635afd55611f13ee7
The seconds and minutes files have always existed in the base language
directory of the Core package. So say.c has always been calling the wrong
location (under digits/) for those two files and in the case of second and
minute they didn't exist in the Core packages at all.
The 1.6 sounds release moves the second and minute files into Core from
Extra for the languages that already had them. A future release will include
the second and minute files for languages that didn't already have them.
This patch just changes all the target locations for second, seconds,
minute, and minutes that were under the digits subdir to be under the root of
sounds instead. Which is where the sounds will be for some languages after 1.6
sounds and for all languages after a future release.
ASTERISK-25810 #close
Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702
Reported-by: Nicolas Riendeau
This change adds support for socket activation of certain SOCK_STREAM
listeners in Asterisk:
* AMI / AMI over TLS
* CLI
* HTTP / HTTPS
Example systemd units are provided. This support extends to any socket
which is initialized using ast_tcptls_server_start, so any unknown
modules using this function will support socket activation.
Asterisk continues to function as normal if socket activation is not
enabled or if systemd development headers are not available during
build.
ASTERISK-27063 #close
Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d
AMI goes from 3.2.0 to 4.0.0
ARI goes from 2.0.0 to 3.0.0
Copied UPGRADE.txt -> UPGRADE-15.txt
Created new UPGRADE.txt
Removed a log file that was accidentally checked in a while ago
Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7
The maximum packet size for PJSIP has been increased to handle the
multiple streams being added for WebRTC.
Change-Id: I9ea1e8d02668c544acadcb1c6200e1cc1bd588b3
When a participant leaves a bridge while operating in SFU mode
their respective stream on every other participant needs to be
removed. Leaving the stream out of the new topology results in
every stream after it being moved and reordered. This causes
problems with clients. Instead simply mark the stream as removed
which leaves it in place in the SDP and doesn't reorder or touch
any other streams.
ASTERISK-27136
Change-Id: I4b3f840adcdf69b83842b0d8a737665ba0ef9cb1
Streams are never truly removed in SDP, they still occupy
a location within the SDP. This location can be reused by
another stream if it so chooses.
This change takes advantage of this such that if a new stream
is needing to be added for a new participant any removed streams
are instead replaced first. This reduces the size of the SDP
and the number of streams.
ASTERISK-27134
Change-Id: I95cdcfd55cf47e02ea52abb5d94008db3fb68b1d
This change makes it so that if an RTCP packet is being sent
the RTP ICE component is used for sending if RTCP-MUX is in use.
ASTERISK-27133
Change-Id: I6200f611ede709602ee9b89501720c29545ed68b
This commit fixes two possible scenarios:
* When recording name and if during recording you hangup, file is never
removed. This is due to the fact file location is nulled.
* When recording name and if you hangup during thank-you prompt, file
is never removed.
ASTERISK-27123 #close
Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625
On every reload of chan_iax2 module, MWI subscription was added, which
results in additional taskprocessors being accumulated over time.
This commit fixes it by making sure we check for existing subscription
first.
This was verified with 'core show taskprocessors' CLI command.
ASTERISK-27122 #close
Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9
This patch creates a new configuration option called "webrtc". When enabled it
defaults and enables the following options that are needed in order for webrtc
to work in Asterisk:
rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
When "webrtc" is enabled, this patch also parses the "msid" media level
attribute from an SDP. It will also appropriately add it onto the outgoing
session when applicable.
Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent.
ASTERISK-27119 #close
Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd
Joshua Colp
Corey Farrell
Sean Bright
George Joseph
Torrey Searle
Jenkins2
Sergej Kasumovic
Matthew Fredrickson
Richard Mudgett
Rusty Newton
Benjamin Keith Ford
Kevin Harwell