You cannot cast a pjsip_uri to a pjsip_sip_uri using pjsip_uri_get_uri,
without checking that it's a PJSIP_URI_SCHEME_IS_SIP(S).
ASTERISK-28936
Change-Id: I9f572b3677e4730458e9402719e580f8681afe2a
In a particular fax gateway scenario whereby it would
have to translate using the read translation path on a
channel the frame being translated would be consumed.
When the frame is in the write path it is not permitted
to free the frame as the caller expects it to continue
to exist.
This change makes it so that the frame is only consumed
on the read path where it is acceptable to free it.
ASTERISK-28900
Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d
When writing tx messages to pcap files, Asterisk is using the wrong
pointer resulting in lots of wasted space. This patch fixes it to use
the correct pointer.
ASTERISK-28932 #close
Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23
Scope tracing isn't supported in Asterisk 13 due to changes made
to logging between 13 and 16 but since the scope tracing macros
may be present in the 16, 17 and master branches, those macros
are defined here as NOOPs so cherry-picking changes downward
to 13 can still be seamless.
Change-Id: I0390ce5651374d8f3e06d7620050acb22c5440a2
The destructor for RTP deallocated transport resources
before terminating the ICE support. This could result
in a crash as the thread handling ICE would access already
freed parts of the RTP data.
This change re-orders the destruction so that ICE is
stopped before destroying things.
ASTERISK-28885
Change-Id: Ie71add549f12b6cdea7e9dbf976d1bd1d2fc0bdc
files
fwrite() does return the number of elements written and not the
number of bytes. However asterisk is currently comparing the return
value to the size of the written element what means that asterisk logs
five WARNING messages on every packet written to the pcap file.
This patch changes the code to check for the correct value, which will
always be 1.
ASTERISK-28921 #close
Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2
When replacing the user portion of the Contact URI the code
was using the ephemeral pool instead of the tdata pool. This
could cause the Contact user value to become invalid after a
period of time.
The code will now use the tdata pool which persists for the
lifetime of the message instead.
ASTERISK-28794
Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5
While asterisk is filtering out the x-ast-orig-host parameter from the
contact on response messages, it is not filtering it out from the
request URI and the to header on SIP requests (for example INVITE).
ASTERISK-28884 #close
Change-Id: Id032b33098a1befea9b243ca994184baecccc59e
When a configuration file in Asterisk is loaded
information about it is stored such that on a
reload it is not reloaded if nothing has changed.
This can be problematic when an error exists in
a configuration file in PJSIP since the error
will be output at start and not subsequently on
reload if the file is unchanged.
This change makes it so that if an error is
encountered when res_sorcery_config is loading
a configuration file a reload will always read
in the configuration file, allowing the error
to be seen easier.
Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed
When receiving audio from a channel we determine if it
is talking or silence based on a threshold value. If
this threshold is met we always mix the audio into the
conference bridge. If this threshold is not met we also
mix the audio into the conference bridge UNLESS the
drop silence option is enabled.
The code that removed the audio from the mixed frame
assumed that it was always not present if it did not
meet the threshold to be considered talking. This is
incorrect. If it has been stated that the audio was
mixed into the mixed frame then it has been mixed into
the mixed frame. By not removing audio that was
considered non-talking it was possible for a channel
to receive a slight echo of audio of itself at times.
This change ensures that the audio is always removed
from the mixed frame going back to the channel so it
no longer receives the slight echo.
ASTERISK-28898
Change-Id: I7b1b582cc1bcdb318ecc60c9d2e3d87ae31d55cb
The PJSIP packet logger now has the following CLI commands:
pjsip set logger pcap <filename>
When used this will create a pcap file containing the incoming
and outgoing SIP packets, in unencrypted form.
pjsip set logger verbose <on / off>
This allows you to toggle logging to verbose on and off.
pjsip set logger host <IP/subnet mask> add
This allows you to add an additional IP address or subnet
mask to logging, allowing you to log multiple instead of
just a single IP address or all traffic.
The normal "pjsip set logger host" CLI command has also been
expanded to allow subnet masks as well.
ASTERISK-28895
Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e
Pjproject makefiles miss some dependencies which can cause race
conditions when building with parallel make processes. This patch
adds such dependencies correctly.
ASTERISK-28879 #close
Reported-by: Dmitry Wagin <dmitry.wagin@ya.ru>
Change-Id: Ie1b0dc365dafe4a84c5248097fe8d73804043c22
Changed source and destination address fields in struct
pjsip_history_entry so that they are long enough to hold an IPv6
address.
ASTERISK-28854
Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e
Ensure that output buffers for the osp_convert_inout
function have sufficient space for additional data
such as brackets and ports.
ASTERISK-28804
Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b
fork before exec
Posix does only allow async-signal-safe syscalls after fork before exec.
As asterisk ignores this, functions like TrySystem or System sometimes
end up in a deadlocked child process. The patch prevents the use of
non-async-signal-safe syscalls.
ASTERISK-28776
Change-Id: Idc76365c0592ee3f3b3bd72a4f48f7a098978e8e
Add a new "masquarade" channel event, and use it in app_queue to track unique id's.
Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210
ASTERISK-28829 #close
ASTERISK-25844 #close
Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6
The configure.m4 script for pjproject contains some += syntax, which
is specific to bash, replacing it with string substitutions makes
the script compatible with traditional Bourne shells.
ASTERISK-28866 #close
Reported-by: Christoph Moench-Tegeder <cmt@FreeBSD.org>
Change-Id: I382a78160e028044598b7da83ec7e1ff42b91c05
The gcc 10 -Wrestrict option was causing "overlap" errors when
snprintf was copying one char[256] structure member to another
char[256] member in the same structure.
Using ast_alloca instead of declaring the structure inline
solves the issue.
Here's a link to the "enhancement":
https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html
We may follow up with a gcc bug report.
Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534
In a Dialplan, the channel drivers 'chan_sip' and 'chan_iax2' support
the channel items 'secure_bridge_media' and 'secure_bridge_signaling'.
That way, a channel can be forced to use encryption even if not
specified in its configuration.
However, when the Local Proxy kicks in, for example, in case of a
forwarding (SIP status 302), Local Proxy stated it does not know those
items. Consequently, such a call could not be proxied how clever your
Dialplan was. Because local calls within Asterisk are always secure,
Local Proxy accepts such a request now.
ASTERISK-22920
Change-Id: I4c143bb70f686790953cc04c5a4b810bbb03636c
In practice it has been seen that some users come
close to our maximum ICE candidate count of 32.
In case people have gone over this increases the
count to 64, giving ample room.
ASTERISK-28859
Change-Id: I35cd68948ec0ada86c14eb53092cdaf8b62996cf
Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined.
ASTERISK-28852 #close
Change-Id: I381718893b80599ab8635f2b594a10c1000d595e
In contrast to RFC 3551, section 4.2, several SIP/RTP clients misbehave
severly (up to crashing). This patch adds another smoother for the audio
received via bt. Therefore the audio frames sent to the core will be
CHANNEL_FRAME_SIZE.
ASTERISK-28832 #close
Change-Id: Ic5f9e2f35868ae59cc9356afbd1388b779a1267f
Fixed it to copy the entire string from the requested endpoint body except tech-prefix.
ASTERISK-28847
Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25
By using pjproject to give us a list of candidates, and then filtering,
if the host has >32 addresses configured, then it is possible that we
end up filtering out all 32 of those, and ending up with no candidates
at all. Instead, get getifaddrs (which pjsip is using underlying
anyway) to retrieve all local addresses, and iterate those, adding the
first 32 addresses not excluded by the ICE ACL.
In our setup at any point in time We've got between 6 and 328 addresses
on any given system. The lower limit is the lower limit but the upper
limit is growing on a near daily basis currently.
Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
The Python/C API is used only if the Test Framework was enabled in Asterisk
'make menuselect'. The Test Framework is available only if the Developer Mode
was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API
is used only if the PJProject was found and not disabled in Asterisk; the user
did not go for './configure --without-pjproject'.
Furthermore, because version 2 of that Python/C API is required (currently) and
because some platforms do not offer a generic version 2, the script searches
for 2.7 explicitly as well.
To avoid version mismatch between the Python/C API and the Python environment,
the script searches for the latter in the same versions, in the same the order
as well. Because this Python/C API is just for (some) Asterisk contributors,
the script also goes for the Python 3 environment as a last resort for all
other Asterisk users. This allows 'make full' even on minimal installations of
Ubuntu 18.04 LTS and newer.
Because the Python/C API is Asterisk contributor specific, the Python packages
are removed from the script './contrib/scripts/install_prereq' as this script
is intended for Asterisk users. Asterisk contributors have to install much more
packages in any case, like:
sudo apt install autoconf automake git git-review python2.7-dev
ASTERISK-28824
ASTERISK-27717
Change-Id: Id46d357e18869f64dcc217b8fdba821b63eeb876
It is possible to configure a TCP/TLS client without having a TCP/TLS
server. In that case, no error or warning was printed but the headers
Contact and Via in SIP REGISTER were "(null)".
ASTERISK-28798
Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2
If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a
digit begin before media, or rtp has been setup then it's possible the
outgoing channel will hear a constant DTMF tone upon answering.
This happens because when there is no media, or rtp chan_pjsip notifies
the core to initiate inband DTMF. However, upon digit end if media, and
rtp become available then chan_pjsip does not notify the core to stop
inband DTMF. Thus the tone continues playing.
This patch makes it so chan_pjsip only notifies the core to start
inband DTMF in only the required cases. Now if there is no media, or
rtp availabe upon digit begin chan_pjsip does nothing, but tells the
core it handled it.
ASTERISK-28817 #close
Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5
