chan_sip: Handle invalid SDP answer to T.38 re-invite
The chan_sip module performs a T.38 re-invite using a single media
stream of udptl, and expects the SDP answer to be the same.
If an SDP answer is received instead that contains an additional
media stream with no joint codec a crash will occur as the code
assumes that at least one joint codec will exist in this
scenario.
Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous
patch introduced a variable of type unassigned long long which is 64-bits.
Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work
with 32-bit systems.
George Joseph [Thu, 27 Jun 2019 17:46:44 +0000 (11:46 -0600)]
pjproject_bundled: Add peer information to most SSL/TLS errors
Most SSL/TLS error messages coming from pjproject now have either
the peer address:port or peer hostname, depending on what was
available at the time and code location where the error was
generated.
Joshua Colp [Mon, 24 Jun 2019 10:08:06 +0000 (10:08 +0000)]
res_pjsip_sdp_rtp: Fix ICE candidates leak.
Given the non-default configuration of enabling ICE support on an
endpoint that does not result in an ICE negotiation occurring the
ICE candidates would be leaked.
This change makes it so that the ICE candidates are only retrieved
if ICE negotiation is occurring.
Alexei Gradinari [Wed, 29 May 2019 22:54:16 +0000 (18:54 -0400)]
res_fax: gateway sends T.38 request to both endpoints if V.21 detected
According T.38 Gateway 'Use case 3'
https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway
T.38 Gateway should send T.38 negotiation request to called endpoint
if FAX preamble (using V.21 detector) generated by called endpoint.
But it does not, because fax_gateway_detect_v21 constructs T.38
negotiation request, but forwards it only to other channel,
not to the channel on which FAX preamble is detected.
Some SIP endpoints could be improperly configured to rely on the other side
to initiate T.38 re-INVITEs.
With this patch the T.38 Gateway tries to negotiate with both sides
by sending T.38 negotiation request to both endpoints supported T.38.
George Joseph [Wed, 19 Jun 2019 16:58:39 +0000 (10:58 -0600)]
CI: New way to determnine libdir
We were using the presence of /usr/lib64 to determine where
shared libraries should be installed. This only existed on
Redhat based systems and was safe. If it existed, use it,
otherwise use /usr/lib.
Unfortunately, Ubuntu 19 decided to create a /usr/lib64 BUT
NOT INCLUDE IT IN THE DEFAULT ld.so.conf. So if anything is
installed there, it won't work.
The new method, just looks for $ID in /etc/os-release and if it's
centos or fedora, uses /usr/lib64 and if ubuntu, uses /usr/lib.
NOTE: This applies only to the CI scripts. Normal asterisk
build and install is not affected.
Alexei Gradinari [Fri, 14 Jun 2019 20:45:39 +0000 (16:45 -0400)]
translate.c do not log WARNING on empty audio frame
There is WARNING "no samples for ..." on each Playtones.
The function ast_playtones_start calls ast_activate_generator,
which calls ast_prod.
The function ast_prod calls ast_write with empty audio frame.
In this case it's spam log.
George Joseph [Mon, 17 Jun 2019 17:11:49 +0000 (11:11 -0600)]
chan_dahdi: Address gcc9 issues
Fixed format-truncation issues in chan_dahdi.c and
sig_analog.c. Since they're related to fields provided
by dahdi-tools we can't change the buffer sizes so we're just
checking the return from snprintf and printing an errior if we
overflow.
George Joseph [Mon, 10 Jun 2019 21:58:59 +0000 (15:58 -0600)]
app_confbridge: Attended transfer event fixup
When a channel already in a conference bridge is attended transfered
to another extension, or when an existing call is attended
transferred into a conference bridge, we now generate ConfbridgeJoin
and ConfbridgeLeave events for the entering and departing channels.
Joshua Colp [Tue, 11 Jun 2019 12:26:42 +0000 (09:26 -0300)]
res_rtp_asterisk: Add support for DTLS packet fragmentation.
This change adds support for larger TLS certificates by allowing
OpenSSL to fragment the DTLS packets according to the configured
MTU. By default this is set to 1200.
This is accomplished by implementing our own BIO method that
supports MTU querying. The configured MTU is returned to OpenSSL
which fragments the packet accordingly. When a packet is to be
sent it is done directly out the RTP instance.
[custom_atxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,AttendedTransfer(${dest})
same => n,Return()
agupta [Thu, 6 Jun 2019 12:48:18 +0000 (18:18 +0530)]
chan_pjsip.c: Check for channel and session to not be NULL in hangup
We have seen some rare case of segmentation fault in hangup function
and we could notice that channel pointer was NULL. Debug log shows
that there is a 200 OK answer and SIP timeout at the same time. It
looks that while the SIP session was being destroyed due to timeout
call hangup due to answer event lead to race condition and channel
is being destroyed from two different places. The check ensures we
check it not to be NULL before freeing it.
[custom_blindxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,BlindTransfer(${dest},default)
same => n,Return()
;;;
Fixes an error occurring in function pgsql_reconnect() caused when value of
hostname is blank. Which in turn will cause the connection string to look
like this: "host= port=xx", which creates a sintax error. This fix now checks
if the corresponding values for host, port, dbname, and user are blank. Note
that since this is a reconnect function the database library will replace any
missing value pairs with default ones.
Alexei Gradinari [Tue, 28 May 2019 20:35:17 +0000 (16:35 -0400)]
res_fax: fix segfault on inactive "reserved" fax session
The change #10017 "Handle fax gateway being started more than once"
introdiced a bug which leads to segfault in res_fax_spandsp.
The res_fax_spandsp module does not support reserving sessions, so
fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE.
The fax_gateway_start does not create a real fax session if the fax session
is already present and the state is not AST_FAX_STATE_RESERVED.
But the "reserved" session created for res_fax_spandsp has state
AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting.
Then when fax_gateway_framehook is called and gateway T.38 state is
NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to
segfault, because session tech_pvt is not set, i.e. the tech session
was not initialized/started.
This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved"
session created for res_fax_spandsp will start.
This patch also adds extra check and log ERROR if tech_pvt is not set
before call tech->write.
Alexei Gradinari [Tue, 28 May 2019 22:15:40 +0000 (18:15 -0400)]
res_fax: add channel name to CLI 'fax show session'
This patch adds a channel name to output of CLI 'fax show session'
and also expands the channel name field up to 30 characters on
CLI 'fax show sessions'
George Joseph [Fri, 17 May 2019 23:44:37 +0000 (17:44 -0600)]
res_rtp_asterisk: Add ability to propose local address in ICE
You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:
Alexei Gradinari [Mon, 13 May 2019 20:37:50 +0000 (16:37 -0400)]
pjsip: replace 180 by 183 if SDP negotiation has completed
The caller endpoint hears dead silence if a callee replies 180 (without SDP)
and the caller already received 183 (with SDP).
It happens because Asterisk sends 180 (WITH SDP) to the caller,
there are not incoming RTP packets from the callee
and Asterisk does not generate inband ringing,
so there are not any outgoing RTP packets to the caller.
This patch replaces 180 by 183 if SDP negotiation has completed,
as if the caller endpoint is configured with "inband_progress=yes".
In this case Asterisk will generate inband ringing untill Asterisk receive
incoming RTP packets from the callee.
Ben Ford [Tue, 7 May 2019 16:08:33 +0000 (11:08 -0500)]
pjsip_options.c: Allow immediate qualifies for new contacts.
When multiple endpoints try to register close together using the same
AOR with qualify_frequency set, one contact would qualify immediately
while the other contacts would have to wait out the duration of the
timer before being able to qualify. Changing the conditional to check
the contact container count for a non-zero value allows all contacts to
qualify immediately.
George Joseph [Fri, 3 May 2019 18:31:06 +0000 (12:31 -0600)]
build: Pass --fno-partial-inlining to third-party when appropriate
When the gcc version is >= 8.2.1, we were already setting the
--fno-partial-inlining flag for Asterisk source files to get around
a gcc bug but we weren't passing the flag down to the bundled
builds of pjproject and jansson.
George Joseph [Thu, 2 May 2019 18:29:49 +0000 (12:29 -0600)]
res_pjsip: Check return from pjsip_parse_uri calls
Updated ast_sip_create_rdata_with_contact and registrar_find_contact
to check the return from pjsip_parse_uri before attempting to
use the uri returned.
ASTERISK-28402 Reported-by: Ross Beer
Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7
Ben Ford [Tue, 23 Apr 2019 14:47:45 +0000 (09:47 -0500)]
stasis: Fix crash at shutdown.
When compiling in dev mode, stasis statistics are enabled and can cause
a crash at shutdown due to the following:
- Containers are freed
- Topics and subscriptions remain
- When those topics and subscriptions are deallocated, they go to do
things with the container
This changes the containers to global ao2 objects, and whenever needed
in the code, a reference must be obtained and checked before any
operations can be done.
Antoni Goldstein [Fri, 29 Mar 2019 14:04:46 +0000 (14:04 +0000)]
app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings
Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.
Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.
Kevin Harwell [Tue, 9 Apr 2019 19:09:44 +0000 (14:09 -0500)]
mwi core: Move core MWI functionality into its own files
There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:
main/mwi.h
main/mwi.c
Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.
George Joseph [Mon, 22 Apr 2019 16:09:51 +0000 (10:09 -0600)]
ARI: Bump non-breaking version number to 1.10.2
main/json.c: Added app_name, app_data to channel type
res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics
res/res_ari: Added timestamp as a requirement for all ARI events
core/buildsystem: check the actual compiler being version
Make compiler check use the output of the actual compiler being
used as reported by the CC variable, instead of unconditionally
running the "gcc" binary. Also only run the check if the compiler
is gcc or a cross-compile gcc.
We changed the validation of autocomplete parameter in the "indications
remove" command to avoid continue the execution of the command after
asking for autocomplete out of range parameters.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
ASTERISK-28375 #close Reported-by: Dan Cropp
Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9
George Joseph [Fri, 12 Apr 2019 16:32:44 +0000 (10:32 -0600)]
CI: Move test group config files to Jenkins
One of the downaides of having things like test configuration
in the git repo is that it can't be changed at runtime. You have
to create a review for the changes and merge it mefore it will
take effect.
This review moves the data currently held in
tests/CI/periodic-dailyTestGroups.json and
tests/CI/gateTestGroups.json into a Jenkins Config File attached
to the job definitions. This allows us to alter it from the
Jenkins UI at runtime. The original files stay in the repo
as documentation.
Sean Bright [Tue, 9 Apr 2019 15:10:12 +0000 (11:10 -0400)]
app_voicemail: Cleanup stale lock files on module load
If Asterisk crashes while a VM directory is locked, lock files in the VM
spool directory will not get properly cleaned up. We now clear them on
module load.
Sean Bright [Thu, 11 Apr 2019 20:48:49 +0000 (16:48 -0400)]
res_ael: Create consistent label names across reloads
Reset the internal counter that the AEL2 compiler uses for unique label
names before compiling. This keeps dialplan labels consistent across
reloads assuming the AEL2 has not changed.
Sean Bright [Thu, 11 Apr 2019 16:03:07 +0000 (12:03 -0400)]
res_ael: Fix pattern matching against literal '+'
When generating the regular expression that matches against existing
extensions, we need to escape literal characters that can also be
regular expression metacharacters. This was already being done for '*'
but we need to do the same for '+'.
In passing, remove some unreachable code - strcmp() is already run
immediately when entering extension_matches().
projects / thirdparty / asterisk.git / log
