The functions that build manager message headers do so in a way that
results in a single messages being split across multiple packets. While
this doesn't matter to the remote end, it makes network captures noisier
and harder to follow, and also means additional system calls.
With this patch, we build up more of the message content into the TLS
buffer before flushing to the network. This change is completely
internal to the manager code and does not affect any of the existing
API's consumers.
Change-Id: I50128b0769060ca5272dbbb5e60242d131eaddf9
Otherwise, OpenR2 threads go crazy and consume almost all CPU resources
Change-Id: I10a41f617613fe7399c5bdced5c64a2751173f28
Signed-off-by: Oron Peled <oron.peled@xorcom.com>
We're at the point where there are enough Jenkins jobs for
Asterisk branches than even cleaned checkouts of Asterisk
will add up to more disk space than is available on the
in-memory workspace mount. Since we archive all relevent
artifacts anyway, there's no need to keep the workspace
around after the job finishes, whether it succeeds or fails.
Change-Id: I1cd3b73ebb045a987df0f62526d152a510210c39
When fixing ASTERISK~24212, a change was done so a scheduled callback could not
be removed while it was running. The caller of ast_sched_del would have to wait.
However, when the caller of ast_sched_del is the callback itself (however wrong
this might be), this new check would cause a deadlock: it would wait forever
for itself.
This changeset introduces an additional check: if ast_sched_del is called
by the callback itself, it is immediately rejected (along with an ERROR log and
a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the
after-ast_sched_del-refcall function is only run if ast_sched_del returned
success.
This should fix the following spurious race condition found in chan_sip:
- thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE)
- thread 2: run sip_poke_peer_now
- thread 2: blank out sched-ID (too soon!)
- thread 1: set sched-ID (too late!)
- thread 2: try to delete the currently running sched-ID
After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor
excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and
other madness) should occur.
(Thanks Richard Mudgett for reviewing/improving this "scary" change.)
Note that this change does not fix the observed race condition: unlocked
access to peer->pokeexpire (and potentially other scheduled items in chan_sip),
causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the
deadlock go away. And in the observed case, it will not have adverse affects
(like memory leaks) because the scheduled item is removed through a different
path.
ASTERISK-28282
Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856
Two new Makefile targets have been added... "install-headers" and
"uninstall-headers" to separately control header installation.
The existing behavior has not changed so "make install" and
"make uninstall" will continue to also install/uninstall the headers.
The new targets were added for forward compatibility with Asterisk 17
in which the headers are no longer installed/uninstalled with the
"install" and "uninstall" targets.
Also corrects an issue where /usr/include/asterisk.h was never
being removed at all.
Change-Id: Ia7399f3a0203a4825fc4a9f43b9034dae9a2b643
When manager debugging is turned on, this patch makes it so incoming AMI actions
are now also logged.
Change-Id: I8047524510e7ac97d99482b2448f8e368f29cd47
We now check that a body exists and it has a length > 0 before
attempting to process it.
ASTERISK-28447
Reported-by: Gil Richard
Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f
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.
This change removes this assumption.
ASTERISK-28465
Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87
Details:
- The memcpy() call copied part of "dahdi_conf" and not "dahdi_conf.mfcr2"
- As a result, the memcmp() in dahdi_r2_get_link() always fails
- This cause dahdi_r2_get_link() to create new link for every channel
(instead of a new link for every ~30 channels)
- With the fix, far less links are generated -- so we use far less threads
Change-Id: I7259dd6272f5e46e8a6c7f5bf3e8c2ec01b8c132
Signed-off-by: Oron Peled <oron.peled@xorcom.com>
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.
ASTERISK-28457
Change-Id: I9cef6b5f2d826fc5c93f2f6a1c997c4e3e6c93fe
Where possble, hostname and port has been added to error
messages, mostly on the server side.
ASTERISK-26006
Reported by: Oleksandr Natalenko
Change-Id: Iff4f897277bc36ce8c5b493b71d0a4a7b74e62f0
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.
ASTERISK-28444
Reported by: Bernhard Schmidt
Change-Id: I41770e8a1ea5e96f6e16b236692c4269ce1ba91e
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.
ASTERISK-28460
Change-Id: I7b3f76f031c41fb8a3dc3ef1a84b77e2a8cb969f
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.
Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39
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.
Change-Id: Iad66374b550fd89349bedbbf2b93f8edd195a7c3
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.
Change-Id: Id4ac309489d9ff281bad02abdef341cecdede660
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.
Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5
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.
Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1
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.
ASTERISK-28018
Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06
Sean Bright
Leonid Fainshtein
Tzafrir Cohen
George Joseph
Friendly Automation
Rodrigo Ramírez Norambuena
Asterisk Development Team
Walter Doekes
Kevin Harwell
Joshua Colp
Francesco Castellano
Chris-Savinovich
Alexei Gradinari