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releases/certified-20.7
master
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${ noResults }
3790 Commits (13.38)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
b87527b000 |
AST-2020-001 - res_pjsip: Return dialog locked and referenced
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 |
5 years ago |
|
|
01b7ac0d59 |
AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit.
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 |
5 years ago |
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715235202d |
pbx.c: On error, ast_add_extension2_lockopt should always free 'data'
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 |
5 years ago |
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9decb33531 |
res_pjsip_diversion: implement support for History-Info
Implemention of History-Info capable of interworking with Diversion
Header following RFC7544
ASTERISK-29027 #close
Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6
(cherry picked from commit
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5 years ago |
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9cd1103030 |
format_cap: Perform codec lookups by pointer instead of name
ASTERISK-28416 #close Change-Id: I069420875ebdbcaada52d92599a5f7de3cb2cdf4 |
5 years ago |
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514b1d6b49 |
Bridging: Use a ref to bridge_channel's channel to prevent crash.
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 |
5 years ago |
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c7df06a077 |
scope_trace: Updated macro stubs to match other branches
Although scope tracing isn't actually implemented in this branch, macro stubs had been added to allow code that used scope tracing to be cherry-picked to this branch without modification. To match the other branches, the following changes were made to the stubs... The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages at the same level as the scope level. This allows the same messages to be printed to the debug log when AST_DEVMODE isn't enabled. Also added a few variants of the SCOPE_EXIT macros that will also call ast_log instead of ast_debug to make it easier to use scope tracing and still print error messages. Change-Id: Ifc281793a5b91c84a4f1633db9d07006d31585ed |
5 years ago |
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179530558d |
bridge_channel: Ensure text messages are zero terminated
T.140 data in RTP is not zero terminated, so when we are queuing a text frame on a bridge we need to ensure that we are passing a zero terminated string. ASTERISK-28974 #close Change-Id: Ic10057387ce30b2094613ea67e3ae8c5c431dda3 |
5 years ago |
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46fb9fb663 |
vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors
The assumed behavior of realloc() - that it was effectively a free() if its second argument was 0 - is Linux specific behavior and is not guaranteed by either POSIX or the C specification. Instead, if we want to resize a vector to 0, do it explicitly. Change-Id: Ife31d4b510ebab41cb5477fdc7ea4e3138ca8b4f |
6 years ago |
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718efe9365 |
utf8.c: Add UTF-8 validation and utility functions
There are various places in Asterisk - specifically in regards to
database integration - where having some kind of UTF-8 validation would
be beneficial. This patch adds:
* Functions to validate that a given string contains only valid UTF-8
sequences.
* A function to copy a string (similar to ast_copy_string) stopping when
an invalid UTF-8 sequence is encountered.
* A UTF-8 validator that allows for progressive validation.
All of this is based on the excellent UTF-8 decoder by Björn Höhrmann.
More information is available here:
https://bjoern.hoehrmann.de/utf-8/decoder/dfa/
The API was written in such a way that should allow us to replace the
implementation later should we determine that we need something more
comprehensive.
Change-Id: I3555d787a79e7c780a7800cd26e0b5056368abf9
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6 years ago |
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4fec6da25b |
vector.h: Add AST_VECTOR_SORT()
Allows a vector to be sorted in-place, rather than only during insertion. Change-Id: I22cba9ddf556a7e44dacc53c4431bd81dd2fa780 |
6 years ago |
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c705e79d79 |
Scope Trace: Update stub defines
Reminder: 13 doesn't support scope tracing but we're keeping the stub defines consistent across branches. Change-Id: If00733a4d5a4e829cbb4c934bcc2f4c1a80dab81 |
6 years ago |
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4bbc5be798 |
manager - Add Content-Type parameter to the SendText action
This patch allows a user of AMI to now specify the type of message content contained within by setting the 'Content-Type' parameter. Note, the AMI version has been bumped for this change. ASTERISK-28945 #close Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb |
6 years ago |
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9fce541643 |
logger: Fix scope trace defines
Although not implemented in the 13 branch, we need to keep the defines consistent across all branches. Change-Id: Id88ad71ca36818a35f1f14a5acb450e240366301 |
6 years ago |
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58eaf41e28 |
pjproject: Upgrade bundled version to pjproject 2.10
This patch makes the usual necessary changes when upgrading to a new version pjproject. For instance, version number bump, patches removed from third-party, new *.md5 file added, etc.. This patch also includes a change to the Asterisk pjproject Makefile to explicitly create the 'source/pjsip-apps/lib' directory. This directory is no longer there by default so needs to be added so the Asterisk malloc debug can be built. This patch also includes some minor changes to Asterisk that were a result of the upgrade. Specifically, there was a backward incompatibility change made in 2.10 that modified the "expires header" variable field from a signed to an unsigned value. This potentially effects comparison. Namely, those check for a value less than zero. This patch modified a few locations in the Asterisk code that may have been affected. Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to check a minimum version of pjproject at compile time. ASTERISK-28899 #close Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81 |
6 years ago |
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654ec4b908 |
Compiler fixes for gcc 10
This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9 |
6 years ago |
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edf9471e05 |
Scope Tracing: Add stubs to allow easier cherry-picking
Scope tracing isn't supported in Asterisk 13 due to changes made to logging between 13 and 16 but since the scope tracing macros may be present in the 16, 17 and master branches, those macros are defined here as NOOPs so cherry-picking changes downward to 13 can still be seamless. Change-Id: I0390ce5651374d8f3e06d7620050acb22c5440a2 |
6 years ago |
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10292dc663 |
app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions
Add a new "masquarade" channel event, and use it in app_queue to track unique id's. Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210 ASTERISK-28829 #close ASTERISK-25844 #close Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6 |
6 years ago |
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947a6e8674 |
func_volume: Accept decimal number as argument
Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c |
6 years ago |
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5ddac396f0 |
acl: implement a centralized ACL output mechanism for HAs and ACLs.
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 |
6 years ago |
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6b2aea9ce2 |
netsock2: compile fixes.
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 |
6 years ago |
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994853658b |
Merge "message & stasis/messaging: make text message variables work in ARI" into 13
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6 years ago |
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c5ababbd2b |
Merge "res/res_pjsip_sdp_rtp: Fix MOH transitions" into 13
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6 years ago |
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0fb06121cb |
message & stasis/messaging: make text message variables work in ARI
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 |
6 years ago |
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ed670ca0c6 |
res/res_pjsip_sdp_rtp: Fix MOH transitions
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 |
6 years ago |
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6af57d5750 |
Merge "say: Remove unused "plural" option from main/say" into 13
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6 years ago |
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2241f828ca |
Merge "format_cap: make function parameters 'const'" into 13
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6 years ago |
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ff87627140 |
Merge "pjsip: Update ACLs on named ACL changes." into 13
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6 years ago |
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c3e9d37ce7 |
say: Remove unused "plural" option from main/say
There are exceptions for plural objects, but they are detected using the supplied NUMBER, not using an extra option. Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113 |
6 years ago |
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d13cd25d4c |
format_cap: make function parameters 'const'
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 |
6 years ago |
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f928600a98 |
RTP/ICE: Send on first valid pair.
When handling ICE negotiations, it's possible that there can be a delay between STUN binding requests which in turn will cause a delay in ICE completion, preventing media from flowing. It should be possible to send media when there is at least one valid pair, preventing this scenario from occurring. A change was added to PJPROJECT that adds an optional callback (on_valid_pair) that will be called when the first valid pair is found during ICE negotiation. Asterisk uses this to start the DTLS handshake, allowing media to flow. It will only be called once, either on the first valid pair, or when ICE negotiation is complete. ASTERISK-28716 Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867 |
6 years ago |
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a60071610c |
pjsip: Update ACLs on named ACL changes.
This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b |
6 years ago |
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bfc93cc954 |
chan_pjsip: Ignore RTP that we haven't negotiated
If chan_pjsip receives an RTP packet whose payload differs from the channel's native format, and asymmetric_rtp_codec is disabled (the default), Asterisk will switch the channel's native format to match that of the incoming packet without regard to the negotiated payloads. We now keep track of the formats that have been negotiated and check before switching payloads which results in these packets being dropped instead of causing the session to terminate. ASTERISK-28139 #close Reported by: Paul Brooks Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3 |
6 years ago |
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0d1fcf0626 |
Merge "netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr." into 13
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6 years ago |
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3e446508b4 |
Merge "netsock2: backport ast_sockaddr_copy_sockaddr to asterisk 13." into 13
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6 years ago |
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e7d2ab666b |
netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr.
ast_addressfamily_to_sockaddrize will determine the size that's required, and ast_sockaddr_from_sockaddr then wraps this new function and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without knowing the address family) into the ast_sockaddr structure. Change-Id: Iee604e96e9096c79b477d6e5ff310cf0b06dae86 Signed-off-by: Jaco Kroon <jaco@uls.co.za> |
6 years ago |
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fa700acf39 |
Merge "res_pjsip_pubsub: Add ability to persist generator state information." into 13
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6 years ago |
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4cefb2f3cb |
Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support" into 13
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6 years ago |
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9720b29d8e |
res_pjsip_pubsub: Add ability to persist generator state information.
Some body generators, such as dialog-info+xml, require storing state information which is then conveyed in the NOTIFY request itself. Up until now there was no way for such body generators to persist this information. Two new API calls have been added to allow body generators to set and get persisted data. This data is persisted out alongside the normal persistence information and allows the body generator to restore state information or to simply use this for normal storage of state. State is stored in the form of JSON and it is up to the body generator to interpret this as needed. The dialog-info+xml body generator has been updated to take advantage of this to persist the version number. ASTERISK-27759 Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de |
6 years ago |
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49bb7d85cd |
res_pjsip_endpoint_identifier_ip.c: Add port matching support
Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92 |
6 years ago |
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e4e5e07e3d |
stasis.c: Use correct topic name in stasis_topic_pool_delete_topic
When a topic is created for an object, its name is only <object>:<uniqueid> For example: bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 When a topic is added to a pool, its name has the pool's topic name prepended. For example: bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 The topic_pool_entry's name however, is only what was passed in to stasis_topic_pool_get_topic which is bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 That's actually correct because the entry is qualified by the pool that's in. When you're ready to delete the entry from the pool, you retrieve the tropic name from the object but since it now has the pool's topic name prepended, it won't be found in the pool container. Fix: * Modified stasis_topic_pool_delete_topic() to skip past the pool topic's name, if it was prepended to the topic name, before searching the container for a pool entry. ASTERISK-28633 Reported by: Joeran Vinzens Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6 |
6 years ago |
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5a3826e334 |
Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" into 13
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6 years ago |
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e8ca2af820 |
netsock2: backport ast_sockaddr_copy_sockaddr to asterisk 13.
This function is already present in 15, 16 and master. Change-Id: I491ddecdccc87620c264567c549217e48f3d4ff1 Signed-off-by: Jaco Kroon <jaco@uls.co.za> |
6 years ago |
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66970af0d3 |
Merge "confbridge: Add support for specifying maximum sample rate." into 13
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6 years ago |
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7cea1c4306 |
configure: Add check for MySQL client bool and my_bool type usage.
Instead of trying to use the defined MySQL client version from the header use a configure check to determine whether the bool or my_bool type should be used for defining a boolean. ASTERISK-28604 Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7 |
6 years ago |
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ca35de4282 |
confbridge: Add support for specifying maximum sample rate.
ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee |
6 years ago |
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754fca9297 |
res_fax: wrap v21 detected Asterisk initiated negotiation with config option
A previous patch: Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38 versus the previous behavior of forwarding negotiation to the "other" channel once a preamble was detected. This had the unfortunate side effect of breaking some setups. Specifically ones that set the max datagram option on an endpoint configuration (configured max datagram was not propagated since Asterisk now initiates negotiations). This patch adds a configuration option, "negotiate_both", that when enabled makes it so Asterisk initiates the negotiation requests to both endpoints vs. the previous behavior of waiting, and forwarding the request. The default is disabled keeping with the old behavior. ASTERISK-28660 Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a |
6 years ago |
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6e79028020 |
ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging.
Due to use in res_rtp_asterisk there is a need to be able to apply an ACL without logging any invalid/denies. It's probably sensible to at least validate the ACL once directly after load and report invalid ACLs. Change-Id: I256169229d945ca7c1bbf228fc492d91df345843 Signed-off-by: Jaco Kroon <jaco@uls.co.za> |
6 years ago |
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6ca76798e1 |
various files - fix some alerts raised by lgtm code analysis
This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25 |
6 years ago |
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f1e3a13579 |
utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN
ASTERISK-28590 #close Change-Id: I51abce00c04d0a06550bda5205580705185b9c1c |
6 years ago |