RFC5621 requires any content type with a Content-Disposition
with handling=required to be rejected with a 415 response
ASTERISK-28782 #close
Change-Id: Iad969df75936730254b95c1a8bc3b48497070bb4
named_acl.c (which is really a named_ha) now uses ast_ha_output.
I've also updated main/manager.c to output the actual ACL on "manager
show user <username>" if one is set. If this works then we can add
similar to other modules as required.
Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f
This patch makes it so ast_coredumper now outputs the following information to
a *-info.txt file when processing a core file:
asterisk version and "built by" string
BUILD_OPTS
system start, and last reloaded date/time
taskprocessor list
equivalent of "bridge show all"
equivalent of "core show channels verbose"
Also a slight modification was made when trying to obtain the pid(s) of a
running Asterisk. If it fails to retrieve any it now reports an error.
Change-Id: I54f35c19ab69b8f8dc78cc933c3fb7c99cef346b
This fixes ast_addressfamily_to_sockaddrsize to reference the
provided argument, and ast_sockaddr_from_sockaddr to not use the name of
a structure as argument.
Change-Id: Ibf5db469c47c3b4214edf8456326086174e8edd7
If a negative (error) return is received from dundi_lookup_internal,
this is not handled correctly when assigning the result to the buffer.
As such, use a signed integer in the assignment and do a proper
comparison.
ASTERISK-21205
Change-Id: I5214ebb6491e2bd14f90c7d3ce229da86888f739
When the Asterisk receives 200 OK with invalid SDP,
the Asterisk/PJPROJECT terminating the session.
But if the channel was in the Bridge, Asterisk tries send
the Re-Invite before terminating the session.
And when the Asterisk sending the Re-Invite, it doesn't check
the SDP is NULL or not. This crashes the Asterisk.
Fixed it to close the session correctly if the UAS sends the
200 OK with wrong SDP.
ASTERISK-28743
Change-Id: Ifa864e0e125b1a7ed2f3abd4164187e1dddc56da
This patch has been included in Gentoo distribution for at least since
asterisk 1.8, but there are references in the logs going back as far as
1.0.0 - not sure if this is still required in any way, it does apply,
and it doesn't (as far as we can determine) cause build failures.
Change-Id: I46d8845e30200205e80580680bf060aa3012ba54
We (Gentoo distribution) reckon that in the case of multiple versions of
gmime installed we should prefer the newest one.
Change-Id: Idf7be613230232eb1d573d93c4a5a8297f4ecd2d
Do not hang up a PJSIP channel on RTP timeout if that channel is in
a direct-media bridge. Also reset the time of the last received RTP packet when
direct-media ends (wait full rtp_timeout period before checking first time after
audio came back to Asterisk).
ASTERISK-28774
Reported-by: Michael Neuhauser
Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1
A pure blacklist is not good enough, we need a whitelist mechanism as
well, and the simplest way to do that is to re-use existing ACL
infrastructure.
This makes it simpler to blacklist say an entire block (/24) except a
smaller block (eg, a /29 or even a /32). Normally you'd need to
recursively split the block, so if you want to blacklist a /24 except
for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I
feel that having an ACL instead of a blacklist only is clearer.
Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
binutils 2.34 merged this commit:
https://sourceware.org/git/gitweb.cgi?p=binutils-gdb.git;a=commitdiff;\
h=fd3619828e94a24a92cddec42cbc0ab33352eeb4
Which effectively does things like:
-#define bfd_section_size(bfd, ptr) ((ptr)->size)
-#define bfd_get_section_size(ptr) ((ptr)->size)
+#define bfd_section_size(sec) ((sec)->size)
So in order to remain backwards compatible we need to detect this API
change, and adjust accordingly. The simplest is to notice that the
bfd_get_section_size and bfd_get_section_vma MACROs are no longer
defined, and define then onto the new API. The alternative is to litter
the code with a number of #ifdef #else #endif splatters right through
the code.
Change-Id: I3efe0f8e8f3e338d16fcbc2b26a505367b6e172f
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 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
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
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
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
