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Attempting to load a transport from realtime was forcing asterisk into an
infinite recursion loop. The first thing transport_apply did was to do a
sorcery retrieve by id for an existing transport of the same name. For files,
this just returns the previous object from res_sorcery_config's internal
container, if any. For realtime, the res_sourcery_realtime driver looks in the
database and finds the existing row but now it has to rehydrate it into a
sorcery object which means calling... transport_apply. And so it goes.
The main issue with loading from realtime (apart from the loop) was that
transport stores structures and pointers directly in the ast_sip_transport
structure instead of the separate ast_transport_state structure. This patch
separates those items into the ast_sip_transport_state structure. The pattern
is roughly the same as res_pjsip_outbound_registration.
Although all current usages of ast_sip_transport and ast_sip_transport_state
were modified to use the new ast_sip_get_transport_state API, the original
items are left in ast_sip_transport and kept updated to maintain ABI
compatability for third-party modules. They are marked as deprecated and
noted that they're now in ast_sip_transport_state.
ASTERISK-25606 #close
Reported-by: Martin Moučka
Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19
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Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98
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A user cannot set new bridge options after the conference is created by
the first user. Attempting to do so is documented as undefined behavior.
This patch ensures that the bridge profile options used are from the
conference and not what a subsequent user may have tried to set.
Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266
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* changes:
app_confbridge: Add ability to get the muted conference state.
app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
app_confbridge: Make non-admin users join a muted conference muted.
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The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
these options, which can cause problems on systems with older OpenSSL
installations.
This commit adds a configure script check for those defines and will not
attempt to make use of those if they do not exist. We will print a
warning urging the user to upgrade their OpenSSL installation if those
defines are not present.
Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d
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ps_systems needed disable_tcp_switch
ps_registrations needed line and endpoint
ASTERISK-25737 #close
Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19
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Device state subscription lifetimes were governed by when the
subscription was established and unsubscribed from. However, it is
possible that at the time of unsubscription, there could be device state
events still in flight. When those device state events occur, the device
state callback could attempt to dereference a freed pointer. Crash.
This change ensures that the lifetime of the device state subscription
does not end until the underlying stasis subscription has confirmed that
its final message has been sent.
Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2
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During ICE negotiation the IPs of the local interfaces are sent to the remote
peer as host candidates. In many cases Asterisk is behind a static one-to-one
NAT, so these host addresses will be internal IP addresses.
To help in hiding the topology of the internal network, this patch adds the
ability to override the host candidates by matching them against a
user-defined list of replacements.
Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f
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This change exposes the configuration of various aspects of the TLS
support and sets the default to the modern standards.
The TLS cipher is now set to the best values according to the
Mozilla OpSec team, different TLS versions can now be disabled, and
the cipher order can be forced to be that of the server instead of
the client.
ASTERISK-24972 #close
Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8
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Sending UDPTL packets to Asterisk with the right amount of missing
sequence numbers and enough redundant 0-length IFP packets, can make
Asterisk crash.
ASTERISK-25603 #close
Reported by: Walter Doekes
ASTERISK-25742 #close
Reported by: Torrey Searle
Change-Id: I97df8375041be986f3f266ac1946a538023a5255
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Setting the sip.conf timert1 value to a value higher than 1245 can cause
an integer overflow and result in large retransmit timeout times. These
large timeout times hold system file descriptors hostage and can cause the
system to run out of file descriptors.
NOTE: The default sip.conf timert1 value is 500 which does not expose the
vulnerability.
* The overflow is now detected and the previous timeout time is
calculated.
ASTERISK-25397 #close
Reported by: Alexander Traud
Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290
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test_dlinklists doesn't need to NOTICE everyone that every macro worked.
res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or
provider was registered.
res_odbc was missing a newline at the end of one message.
Change-Id: I6c06361518ef3711821795e535acd439782a995e
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asterisk's own ip."
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A regression was introduced where searching for realtime PJSIP objects
by regex by starting the regex with a leading "^" would cause no items
to be returned.
This was due to a change which attempted to drop the requirement for a
leading "^" to be present due to how some CLI commands formulate their
regexes. However, the change, rather than simply eliminating the
requirement, caused any regexes that did begin with "^" to end up not
returning the expected results.
This change fixes the problem by inspecting the regex and formulating
the realtime query differently depending on if it begins with "^".
ASTERISK-25702 #close
Reported by Nic Colledge
Patches:
realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691
Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693
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The module res_xmpp does not accept usernames in the form used in component
mode (XEP-0114). In component mode there is no @something in the name.
In component mode the connection is now not dropped anymore.
If the xmpp server sends out a "stream" tag before handshake is finished,
the connection gets dropped in res_xmpp. Now this tag will be ignored and
the connection will be established.
After connecting there will be an exchange of presence states. This does
not work as expected in component mode. The responsible function
"xmpp_pak_presence" is left before the states get sent out. Sending
presence states in component mode is now moved to the top of the function.
ASTERISK-25735 #close
Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c
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Fix some warnings found with clang.
Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd
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A recent commit set qualify_timeout to Decimal which isn't supported.
This path corrects it to Float.
Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf
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When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a)
AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect
asterisk to include the same value for its own ip in both cases a) and b),
but it seems a) produces a contact header like Contact:
<sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like
<sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf
My guess is that manager_sipnotify should call
ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does,
because after applying this patch, both cases a) and b) produce
the contact header that I expect: <sip:asterisk@192.168.1.227:8060>
Reported by: Stefan Engström
Tested by: Stefan Engström
Change-Id: I86af5e209db64aab82c25417de6c768fb645f476
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missing"
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Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4
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Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51
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* Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.
* Added Muted header to AMI ConfbridgeListRooms action response list
events to indicate the muted conference state.
* Added Muted column to CLI "confbridge list" output to indicate the muted
conference state and made the locked column a yes/no value instead of a
locked/unlocked value.
ASTERISK-20987
Reported by: hristo
Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1
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Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7
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ASTERISK-20987 #close
Reported by: hristo
Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38
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Since res_pjsip now depends on res_pjproject, this has been added to
basic-pbx modules.conf.
Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d
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The Makefile only optionally includes makeopts so when goals like uninstall that
dont depend on anything else are run after a distclean, rules like
'rm -f "$(DESTDIR)$(ASTMODDIR)/"*' get run as 'rm -f ""/*' which attempts
to remove everything in the root directory.
Although there's a rule defined for makeopts which prints a message and does
an 'exit 1', since '-include makepopts' was specified (with the -), the exit
was ignored letting the rest of the rules run.
This patch makes makeopts required unless the goal has the string 'clean' in it.
ASTERISK-25730 #close
Reported-by: George Joseph
Change-Id: I1bce59a7ea4f48e7a468e22b2abbb13c63417ac7
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The config options framework is strict in that configuration options must
be documented unless XML documentation support is not available. In
practice this is useful as it ensures documentation exists however in
off-nominal cases this can cause strange problems.
If it is expected that a config option has a non-zero or non-empty
default value but the config option documentation is unavailable
this reasonable expectation will not be met. This can cause obscure
crashes and weirdness depending on how the code handles it.
This change tweaks the behavior to ensure that the config option
is still allowed to register, apply default values, and be set when
devmode is not enabled. If devmode is enabled then the option can
NOT be set.
This also does not remove the initial documentation error message that
is output on load when registering the configuration option.
ASTERISK-25725 #close
Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8
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A recent change to queue channel variable setting to the Stasis control
queue caused a regression. When setting channel variables, it is
possible to give a NULL channel variable value in order to unset the
variable (i.e. remove it from the channel variable list). The change
introduced a call to ast_variable_new(), which is not tolerant of NULL
channel variable values.
This new change switches from using ast_variable to using a custom
channel variable struct that is lighter weight and NULL value-tolerant.
Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d
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Core and extra sounds 1.5 was recently released! The tarballs contain
change descriptions however I figure more people will see this one so
I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra
to Core for en, en_GB, fr and added for languages that didn't already
have Extra sound sets (it,ja,ru).
In addition all of the English and Russian sounds have been completely
re-recorded.
Sounds moved and added:
activated,added,all-circuits-busy-now,astcc-followed-by-pound
at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy
,call-fwd-unconditional,calling,call-waiting,cancelled,
cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated
,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist
,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello
,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to
,location,number,number-not-answering,num-was-successfully,one-moment-please
,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option
,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached
,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial
,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension
,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered
,your
There were also a few random fixes here and there to file names for a few
of the languages.
ASTERISK-25068 #close
Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3
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A test recently uncovered that running an ill-timed AMI command to show
inbound subscriptions could cause a crash since Asterisk will try to
operate on a freed subscription.
The fix for this is to remove the subscription tree from the list of
subscriptions at the time that we are sending our final NOTIFY request
out. This way, as the subscription is in the process of dying, it is
inaccessible from AMI.
Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23
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sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer
of 256 characters. This patch reduces the copy to 255 characters to leave
room for the string null terminator.
ASTERISK-25722 #close
Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab
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