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res_stun_monitor will fail to load if DNS resolution of the STUN server
fails. Instead, we continue without the STUN server being resolved and
we will re-attempt the resolution on the STUN refresh interval.
ASTERISK-21856 #close
Reported by: Jeremy Kister
Change-Id: I6334c54a1cc798f8a836b4b47948e0bb4ef59254
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Sun's Au file format has a minimum data offset 24 bytes, but this
offset is encoded in each .au file. Instead of assuming the minimum,
read the actual value and store it for later use.
ASTERISK-20984 #close
Reported by: Roman S.
Patches:
asterisk-1.8.20.0-au-clicks-2.diff (license #6474) patch
uploaded by Roman S.
Change-Id: I524022fb19ff2fd5af2cc2d669d27a780ab2057c
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Change-Id: Iac40ecb20e10513d67bf0eaf61807f306067b258
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This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compact_headers=yes via the file pjsip.conf.
ASTERISK-26932 #close
Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689
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Change-Id: If99e3b4fc2d7e86fc3e61182aa6c835b407ed49e
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In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
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When a T.38 happens immediatly after call establishment, the control
frame can be lost because the other leg is not yet in the bridge.
This patch detects this case an makes sure T.38 negotation happens
when the 2nd leg is being made compatible with the negotating
first leg
ASTERISK-26923 #close
Change-Id: If334125ee61ed63550d242fc9efe7987e37e1d94
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On 2's compliment machines abs(INT_MIN) behavior is undefined and
results in a negative value still being returnd. This results in
negative hash codes that can result in crashes.
ASTERISK-26528 #close
Change-Id: Idff550145ca2133792a61a2e212b4a3e82c6517b
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Added the stun_blacklist option to rtp.conf. Some multihomed servers have
IP interfaces that cannot reach the STUN server specified by stunaddr.
Blacklist those interface subnets from trying to send a STUN packet to
find the external IP address. Attempting to send the STUN packet
needlessly delays processing incoming and outgoing SIP INVITEs because we
will wait for a response that can never come until we give up on the
response. Multiple subnets may be listed.
ASTERISK-26890 #close
Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342
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If ast_stun_request() receives packets other than a STUN response then we
could conceivably never exit if we continue to receive packets with less
than three seconds between them.
* Fix poll timeout to keep track of the time when we sent the STUN
request. We will now send a STUN request every three seconds regardless
of how many other packets we receive while waiting for a response until we
have completed three STUN request transmission cycles.
Change-Id: Ib606cb08585e06eb50877f67b8d3bd385a85c266
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Return early if ast_sorcery_retrieve_by_id() is not passed an id to find.
Also eliminated the RAII_VAR() usage in the function.
Change-Id: I871dbe162a301b5ced8b4393cec27180c7c6b218
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Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1
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* create_rtp(): Eliminate use of deprecated transport struct member. That
member and several others in the transport structure were deprecated
because of an infinite loop created when using realtime configuration.
See 2451d4e4550336197ee2e482750cc53f30afa352
ASTERISK-26851
Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc
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Temporarily running out of file descriptors should not terminate the
listener thread. Otherwise, when there becomes more file descriptors
available, nothing is listening.
* Added EMFILE exception to abnormal thread exit.
* Added an abnormal TCP/TLS listener exit error message.
* Closed the TCP/TLS listener socket on abnormal exit so Asterisk does not
appear dead if something tries to connect to the socket.
ASTERISK-26903 #close
Change-Id: I10f2f784065136277f271159f0925927194581b5
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This include was accidentally removed in changeset
Ia79aea64de89531362e993e34230c2044a70aa93. My bad.
Change-Id: I1d716c7f9590b4e97909fb8bca1f2ed9bd0e4082
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ioqueue_on_read_complete()."
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This change adds database tables for the PUBLISH support so it
can be configured using realtime. A minor fix to the
res_pjsip_publish_asterisk module was done so that it read the
sorcery configuration from the correct section. Finally the
sample configuration files have been updated.
ASTERISK-26928
Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952
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When the Asterisk channel driver res_pjsip offers SIP-over-TLS, sometimes, not
reproducible, Asterisk crashed in pj_ssl_sock_get_info() because a NULL pointer
was read. This change avoids this crash.
ASTERISK-26927 #close
Change-Id: I24a6011b44d1426d159742ff4421cf806a52938b
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This takes care of warnings by ossobv/asterisklint.
Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93
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0035-r5572-svn-backport-dialog-transaction-deadlock.patch
0036-r5573-svn-backport-ua-pjsua-transaction-deadlock.patch
0037-r5576-svn-backport-session-timer-crash.patch
Also removed the progress bar from wget download to stdout.
ASTERISK-26905 #close
Reported-by: Ross Beer
Change-Id: I268fb3cf71a3bb24283ff0d24bd8b03239d81256
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We needed the reason for our reporting when agents pause/unpause all of
their queues at once. This is a small, simple patch that adds a reason
for PAUSEALL and UNPAUSEALL. I have been using it in production for years.
ASTERISK-26920 #close
Change-Id: Ifb3f0d1a0abd5194253d9794023546e1395baf3d
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Added:
* outbound-publish
* resource_list
* inbound-publication
* asterisk-publication
Change-Id: I65043a896c35483f30a92d30b5b118359af7ba5a
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* create_rtp(): Fix unexpected alteration of global address_rtp if a
transport is bound to an address.
* create_rtp(): Fix use of uninitialized memory if the endpoint RTP media
address is invalid or the transport has an invalid address.
ASTERISK-26851
Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7
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ast_cdr_setuserfield wrote to a fixed length field using strcpy. This could
result in a buffer overrun when called from chan_sip or func_cdr. This patch
adds a maximum bytes written to the field by using ast_copy_string instead.
ASTERISK-26897 #close
patches:
0001-CDR-Protect-from-data-overflow-in-ast_cdr_setuserfie.patch submitted
by Corey Farrell (license #5909)
Change-Id: Ib23ca77e9b9e2803a450e1206af45df2d2fdf65c
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Purge Realtime MOH classes on 'moh reload' even when musiconhold.conf
hasn't changed.
ASTERISK-25974 #close
Change-Id: I42c78ea76528473a656f204595956c9eedcf3246
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* Report failures if configure finds a required header is missing.
* Deduplicate includes between asterisk.h, astmm.h and compat.h.
* Unconditionally include headers in compat.h if required elsewhere.
Change-Id: Ie67d0185ca71fbfb81c9bdfaebe46a49e3c56dc5
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We were leaking a transport ref in multihomed_on_rx_message() which
resulted in the FRACK about excessive ref counts.
ASTERISK-26916 #close
Change-Id: I7a96658a9614a060565bb9ad51cb1c9c11ee145f
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SIP user-agents indicate which protocol extensions are allowed in headers
like Supported and Required. Such protocol extensions are Session Timers
(RFC 4028) for example. Session Timers are supported since Mantis-10665.
Since ASTERISK-21721, not only the first but multiple Supported/Required
headers in a message are parsed. In that change, an existing variable was
re-used within a newly added do-loop. Currently, at the end of that loop,
that variable is an empty string always. Previously, that variable was used
within log output. However, the log output was not changed.
ASTERISK-26915 #close
Change-Id: I09315f31b4d78fb214bb2a9fb6c0f5e143eae990
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It is perfectly acceptable for a BYE to be sent on a disconnected
session. This occurs when we respond to a challenge to the BYE
for authentication credentials.
ASTERISK-26363
Change-Id: I6ef0ddece812fea6665a1dd2549ef44fb9d90045
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Added missing channel technology read/write stream callback
initialization.
Change-Id: I829043a327d987e0d964485dd3d27964bebbd623
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