If an SSL socket parent/listener was destroyed during the handshake,
depending on timing, it was possible for the handling callback to
attempt access of it after the fact thus causing a crash.
ASTERISK-29415 #close
Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d
If chan_iax2 received a packet with an unsupported media format, for
example vp9, then it would set the frame's format to NULL. This could
then result in a crash later when an attempt was made to access the
format.
This patch makes it so chan_iax2 now ignores/drops frames received
with unsupported media format types.
ASTERISK-29392 #close
Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1
New responses sent within a PJSIP sessions are based on those that were
sent before. Therefore, adding/modifying a header once causes it to be
sent on all responses that follow.
Sending 181 Call Is Being Forwarded many times first adds "histinfo"
duplicated more and more, and eventually overflows past the array
boundary.
This commit adds a check preventing adding "histinfo" more than once,
and skipping it if there is no more space in the header.
Similar overflow situations can also occur in res_pjsip_path and
res_pjsip_outbound_registration so those were also modified to
check the bounds and suppress duplicate Supported values.
ASTERISK-29227
Reported by: Ivan Poddubny
Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322
If a remote side is broken and sends an SDP that can not be
negotiated the call will be torn down but there is a window
where a second 183 Session Progress or 200 OK that is forked
can be received that also attempts to negotiate SDP. Since
the code marked the SDP negotiation as being done and complete
prior to this it assumes that there is an active local and remote
SDP which it can modify, while in fact there is not as the SDP
did not successfully negotiate. Since there is no local or remote
SDP a crash occurs.
This patch changes the pjmedia_sdp_neg_modify_local_offer2
function to no longer assume that a previous SDP negotiation
was successful.
ASTERISK-29196
Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e
Add a check to see if the URI is a Tel URI and prevent crashing on
trying to retrieve the reason parameter.
ASTERISK-29191
ASTERISK-29219
Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37
pjproject returns the dialog locked and with a reference. However,
in Asterisk the method that handles this decrements the reference
and removes the lock prior to returning. This makes it possible,
under some circumstances, for another thread to free said dialog
before the thread that created it attempts to use it again. Of
course when the thread that created it tries to use a freed dialog
a crash can occur.
This patch makes it so Asterisk now returns the newly created
dialog both locked, and with an added reference. This allows the
caller to de-reference, and unlock the dialog when it is safe to
do so.
In the case of a new SIP Invite the lock, and reference are now
held for the entirety of the new invite handling process.
Otherwise it's possible for the dialog, or its dependent objects,
like the transaction, to disappear. For example if there is a TCP
transport error.
ASTERISK-29057 #close
Change-Id: I5ef645a47829596f402cf383dc02c629c618969e
If Asterisk sends out an INVITE and receives a challenge with a
different nonce value each time, it will continuously send out INVITEs,
even if the call is hung up. The endpoint must be configured for
outbound authentication for this to occur. A limit has been set on
outbound INVITEs so that, once reached, Asterisk will stop sending
INVITEs and the transaction will terminate.
ASTERISK-29013
Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7
RFC 8760 added new digest-access-authentication schemes. Testing
revealed that chan_sip does not pick MD5 if several schemes are offered
by the User Agent Server (UAS). This change does not implement any of
the new schemes like SHA-256. This change makes sure, MD5 is picked so
UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can
still be used. This should have worked since day one because SIP/2.0
already envisioned several schemes (see RFC 3261 and its augmented BNF
for 'algorithm' which includes 'token' as third alternative; note: if
'algorithm' was not present, MD5 is still assumed even in RFC 7616).
Change-Id: I61ca0b1f74b5ec2b5f3062c2d661cafeaf597fcd
Version: gcc (Ubuntu 9.3.0-10ubuntu2) 9.3.0
Warning:
say.c:2371:24: error: ‘%d’ directive output may be truncated writing
between 1 and 11 bytes into a region of size 10
[-Werror=format-truncation=]
2371 | snprintf(buf, 10, "%d", num);
say.c:2371:23: note: directive argument in the range [-2147483648, 9]
That's not possible though, as the if() starts out checking for (num < 0),
making this Warning a false positive.
(Also replaced some else<TAB>if with else<SP>if while in the vicinity.)
Change-Id: Ic7a70120188c9aa525a6d70289385bfce878438a
This patch initializes a couple of local variables to some default values.
Interestingly, in the 'pj_status_t dlg_status' case the value not being
initialized caused memory to grow, and not be recovered, in the off nominal
path (at least on my machine).
Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7
Ubuntu 20.10 does not come with GMime 2.6. Ubuntu 16.04 LTS does not
come with GMime 3.0. aptitude ignores any missing package. Therefore,
it installs the correct package(s). However, in Ubuntu 18.04 LTS and
Ubuntu 20.04 LTS, both versions are installed alongside although only
one is really needed.
Change-Id: Ic58aa9f2e131d94671f286f17dbd61e1ccbabcb7
Note to maintainers: Lua 5.4, Lua 5.3, and Lua 5.2 have not been tested
at runtime with pbx_lua. Until then, use the lowest available version
of Lua, if you enabled the module pbx_lua at all.
Change-Id: Ie5270448b11fcb4e2a53d899e4fe7fea793ce7e0
This change outputs a message at startup and when a remote
console is connected stating that this branch is no longer
receiving bug fixes and to consult the Asterisk Versions wiki
page for status information.
Change-Id: I6b8a77d33e338a3e52dd65f998d8a07e4d16aa2e
When handling a send_message request to a non-existing endpoint, the response's
body is overriden and not properly freed.
ASTERISK-29108
Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8
In the event that the desired extension already exists,
ast_add_extension2_lockopt() will free the 'data' it is passed before
returning an error, so we should not be freeing it ourselves.
Additionally, there were two places where ast_add_extension2_lockopt()
could return an error without also freeing the 'data' pointer, so we
add that.
ASTERISK-29097 #close
Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae
Stop advertising RFC2833 support on the rtp_engine when DTMF mode is
auto but no tel_event was found inside SDP file.
On an incoming call create_rtp will be called and when session->dtmf is
set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without
looking at the SDP file.
Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND
but continued to advertise RFC2833 support.
This meant the native_rtp bridge would falsely consider the two channels
as compatible. In addition to changing the DTMF mode we now set or
remove the AST_RTP_PROPERTY_DTMF.
The property is checked in ast_rtp_dtmf_compatible and called by
native_rtp_bridge_compatible.
ASTERISK-29051 #close
Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287
Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and
AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice,
resulting in to 181 being generated.
Change-Id: I866e5461564644ffb8a5e12b6f1330b50a7b63ab
Only track our sample offset if we are playing a non-announcement file,
otherwise we will skip that number of samples when we start playing the
first MoH file.
ASTERISK-24329 #close
Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc
Sometimes not play MOH on bridge.
ASTERISK-29081
Reported-by: Michal Hajek <michal.hajek@daktela.com>
Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232
Because we use shared thread-local cURL instances, we need to ensure
that the state of the cURL instance is correct before each invocation.
In the case of custom headers, we were not resetting cURL's internal
HTTP header pointer which could result in a crash if subsequent
requests do not configure custom headers.
ASTERISK-29085 #close
Change-Id: I8b4ab34038156dfba613030a45f10e932d2e992d
Implemention of History-Info capable of interworking with Diversion
Header following RFC7544
ASTERISK-29027 #close
Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6
(cherry picked from commit 5a12463c07)
Currently, the ps_contacts table's reg_server column in realtime database type is varchar(20).
This is fine for normal cases, but if the hostname is longer than 20, it returns error and then
failed to register the contact address of the peer.
Normally, 20 characters limitation for the hostname is fine, but with the cloud env.
So, increased the size to 255.
ASTERISK-29056
Change-Id: Iac52c8c35030303cfa551bb39f410b33bffc507d
Currently, it was not possible to create bridge with video_mode single.
This made hard to put the bridge in a vidoe_single mode.
So, added video_single option for Bridge creation using the ARI.
This allows create a bridge with video_mode single.
ASTERISK-29055
Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae
There's a race condition with bridging where a bridge can be torn down
causing the bridge_channel's ast_channel to become NULL when it's still
needed. This particular case happened with attended transfers, but the
crash occurred when trying to publish a stasis message. Now, the
bridge_channel is locked, a ref to the ast_channel is obtained, and that
ref is passed down the chain.
Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814
Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again.
Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6
This fixes a bug introduced mistakenly in ASTERISK-25665:
If leave-empty is enabled, a call may sometimes be removed from
a queue without recording it as abandoned.
This causes Asterisk to not generate an abandon event for that
call, and for the queue abandoned counter to be incorrect.
ASTERISK-29043 #close
Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7
If you run ast_coredumper --tarball-coredumps in the same directory
as the actual coredump, tar can fail because the link to the
actual coredump becomes recursive. The resulting tarball will
have everything _except_ the coredump (which is usually what
you need)
There's also an issue that the directory name in the tarball
is the same as the coredump so if you extract the tarball the
directory it creates will overwrite the coredump.
So:
* Made the link to the coredump use the absolute path to the
file instead of a relative one. This prevents the recursive
link and allows tar to add the coredump.
* The tarballed directory is now named <coredump>.output instead
of just <coredump> so if you expand the tarball it won't
overwrite the coredump.
Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea
When fixing issues uncovered by GCC10 a copy of the parker UUID
was removed accidentally. This change restores it so that the
subscription has the data it needs.
ASTERISK-29042
Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a
Add the option 'enable_callee_prompt' to followme.conf. Enabled by
default. If disabled, a callee is not prompted to accept or reject
the forwarded call.
ASTERISK-29010 #close
Change-Id: Ic15a2bac4f16d0fce7a1b5e7b375f9bafee37aa4
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
The name of the voicemail context was overwriting the name of the
subscribed mailbox. Fix by simplifying how we create the MWI
subscription.
ASTERISK-29029 #close
Change-Id: Ie8a7db6a0b68f3995b0846bbb733a21909ba44e5
With the latest Linux, 'ifconfig' is not installed on default anymore.
Furthermore, the output of the current net-tools 'ifconfig' changed.
Therefore, parsing failed. This update uses 'ip addr show' instead.
Finally, the service for the external IP changed.
Change-Id: I9b1a7c3f457e3553b50a3e9a55524e40d70245a0
Since ASTERISK_27978 the default is not off but 90 seconds. That change
happened because ASTERISK_27347 disabled the keep-alives in the bundled
PJProject and Asterisk should behave the same as before.
Change-Id: Ie63dc558ade6a5a2b969c30a4bd492d63730dc46
This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function
is called on a channel prior to answering a warning is issued and the
function returns unsuccessful.
ASTERISK-28878 #close
Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb
When the ExtensionState AMI action is executed on a pattern matched
hint it can end up adding a new hint if one does not already exist.
This results in a locking order of contexts -> hints -> contexts.
If at the same time a reload is occurring and adding its own hint
it will have a locking order of hints -> contexts.
This results in a deadlock as one thread wants a lock on contexts
that the other has, and the other thread wants a lock on hints
that the other has.
This change enforces a hints -> contexts locking order by explicitly
locking hints in the places where a hint is added when queried for.
This matches the order seen through normal adding of hints.
ASTERISK-29046
Change-Id: I49f027f4aab5d2d50855ae937bcf5e2fd8bfc504
Properly bump reference on format object to avoid memory corruption on double free
ASTERISK-29040 #close
Change-Id: Ic5a7faabfe2ef965ddb024186e1de7ca4542e2a3
Adapt the response handler so it also called when 181 is received.
In the case 181 is received, also generate the 181 response.
ASTERISK-29001 #close
Change-Id: I73cfee46a8ca85371280ebdb38674f8fde7510df
Rather than putting messages into INBOX and then moving them to Urgent
later, put them directly in to the Urgent folder. This prevents
mailcmd from being skipped.
ASTERISK-27273 #close
Change-Id: I49934e093290d308506ab8d45a40ef705c5ae4f5
Asterisk Development Team
Kevin Harwell
Alexander Traud
Ivan Poddubnyi
Joshua C. Colp
Torrey Searle
Kevin Harwell
Ben Ford
Walter Doekes
Jean Aunis
Sean Bright
Holger Hans Peter Freyther
Michal Hajek
Sungtae Kim
Patrick Verzele
Kfir Itzhak
George Joseph
Tzafrir Cohen
Nickolay Shmyrev