mirror of https://github.com/asterisk/asterisk
master
22
23
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10.3.0
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certified/1.8.11-cert1
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certified/11.6-cert6
certified/11.6-cert7
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certified/13.1-cert2
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${ noResults }
4899 Commits (cf637f2510d5e69855d62a4c3306094abfcbb68c)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
200493d9cc |
Merge "tcptls: Avoiding ERR_remove_state in OpenSSL."
|
11 years ago |
|
|
f451af65c4 |
Merge topics 'ASTERISK-25049', 'ASTERISK-25056'
* changes: CLI: Enable automatic references to modules. Modules: Make ast_module_info->self available to auxiliary sources. |
11 years ago |
|
|
7462207694 |
Merge "features: Fix crash when transferee hangs up during DTMF attended transfer."
|
11 years ago |
|
|
d2e2271874 |
manager: Fix build due to missing variable usage.
Change-Id: I26d4d2cb9cee924632ff59ef0b30a7e6a1e2b00d |
11 years ago |
|
|
6c4d1c3223 |
features: Fix crash when transferee hangs up during DTMF attended transfer.
A crash happens with this sequence of steps: 1) Party A is connected to party B. 2) Party B starts a DTMF attended transfer. 3) Party A hangs up while party B is dialing party C. When party A hangs up the bridge that party A and party B are in is dissolved and party B is kicked out of the bridge. When party B finishes dialing party C he attempts to move to the new bridge with party C. Since party B is no longer in a bridge the attempted move dereferences a NULL bridge_channel pointer and crashes. * Made the hold(), unhold(), ringing(), and the bridge_move() functions tolerant of the channel not being in a bridge. The assertion that party B is always in a bridge is not true if the bridged peer of party B hangs up and dissolves the bridge. Being tolerant of not being in a bridge allows the peer hangup stimulus to be processed by the FSM. * Made the bridge_move() function return void since where the return value for a failed move was checked generated a FSM coding ERROR message for a normal off-nominal condition. * Eliminated most uses of RAII_VAR in bridge_basic.c. ASTERISK-25003 #close Reported by: Artem Volodin Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada |
11 years ago |
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bc8dcbdfbc |
Merge "stasis: Fix dial masquerade datastore lifetime"
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11 years ago |
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f36f271c8e |
Merge "vector: Traversal, retrieval, insert and locking enhancements"
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11 years ago |
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247fef6653 |
tcptls: Avoiding ERR_remove_state in OpenSSL.
ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were called by SSL_load_error_strings already and got removed. These changes allow OpenSSL forks like BoringSSL to be used with Asterisk. ASTERISK-25043 #close Reported by: Alexander Traud patches: asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520) Change-Id: If1c0871ece21a7e0763fafbd2fa023ae49d4d629 |
11 years ago |
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f45833c9ad |
Merge "Restrict functionality when ACLs are misconfigured."
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11 years ago |
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11f650c6ac |
stasis: Fix dial masquerade datastore lifetime
A recent change went into Asterisk which added reference counts to the channels stored in a dial masquerade datastore. Unfortunately this included a reference to the caller in a dialing operation. While all of the dialed targets have the datastore removed from them upon dialing completion this did not occur for the caller, causing it to have a reference to itself that could go never go away (as it depended on the destruction of the datastore which only happened when the channel was destroyed). This resulted in the caller channel remaining on the system despite it having hung up. This change does the following to fix this issue: 1. The dial masquerade datastore is now removed from the caller upon dialing completion, just like the dialed targets. 2. Upon destruction of the caller all the dialed targets are also removed from the dial masquerade datastore (just in case). 3. The reference to the caller has been removed as it should not be possible for the datastore to now be valid/useful after the lifetime of the caller has ended. ASTERISK-25025 #close Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f |
11 years ago |
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df6c1d755f |
CLI: Enable automatic references to modules.
* Pass module to ast_cli_register and ast_cli_register_multiple. * Add a module reference before executing any CLI callback, remove the reference when complete. ASTERISK-25049 #close Reported by: Corey Farrell Change-Id: I7aafc7c9f2b912918f28fe51d51e9e8a755750e3 |
11 years ago |
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a8bfa9e104 |
Modules: Make ast_module_info->self available to auxiliary sources.
ast_module_info->self is often needed to register items with the core. Many modules have ad-hoc code to make this pointer available to auxiliary sources. This change updates the module build process to make the needed information available to all sources in a module. ASTERISK-25056 #close Reported by: Corey Farrell Change-Id: I18c8cd58fbcb1b708425f6757becaeca9fa91815 |
11 years ago |
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6d5941297b |
vector: Traversal, retrieval, insert and locking enhancements
Renamed AST_VECTOR_INSERT to AST_VECTOR_REPLACE because it really does replace not insert. The few users of AST_VECTOR_INSERT were refactored. Because these are macros, there should be no ABI compatibility issues. Added AST_VECTOR_INSERT_AT that actually inserts an element into the vector at a specific index pushing existing elements to the right. Added AST_VECTOR_GET_CMP that can retrieve from the vector based on a user-provided compare function. Added AST_VECTOR_CALLBACK function that will execute a function for each element in the vector. Similar to ao2_callback and ao2_callback_data functions although the vector callback can take a variable number of arguments. This should allow easy migration to a vector where a container might be too heavy. Added read/write locked vector and lock manipulation macros. Added unit tests. ASTERISK-25045 #close Change-Id: I2e07ecc709d2f5f91bcab8904e5e9340609b00e0 |
11 years ago |
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4f4aaa0c30 |
main/test.c: Add test to verify there were no registration errors.
This adds a test that will fail if any test failed to register. Also fail if any test registration produced a warning about missing a leading or trailing slash. ASTERISK-25053 #close Reported by: Corey Farrell Change-Id: I93e50b8fcbcfa7f1f5b41b2c44a51685c09529c3 |
11 years ago |
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465bb42cc0 |
Merge "Format Interfaces: Prevent unload except by shutdown."
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11 years ago |
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0aab8cdcb4 |
Merge "main/presencestate.c: Add trailing slash to test category."
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11 years ago |
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12809721d1 |
Merge "Remove unneeded uses of optional_api providers."
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11 years ago |
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981084f08c |
Format Interfaces: Prevent unload except by shutdown.
Format interfaces cannot be unregistered, so the modules that provide them need to be held open except by shutdown. ASTERISK-25054 #close Reported by: Corey Farrell Change-Id: Iadbd9675bf0d30b8fded5a739b163db3ea2db8f3 |
11 years ago |
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1368dae773 |
main/presencestate.c: Add trailing slash to test category.
ASTERISK-25053 Reported by: Corey Farrell Change-Id: I8c0375dd0818747b2d2e1ceaea87bfbeb2daf8d4 |
11 years ago |
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25b67ed6ca |
Merge "term: send proper reset sequence when black background is forced"
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11 years ago |
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c3ec5da156 |
Remove unneeded uses of optional_api providers.
A few cases exist where headers of optional_api provders are included but not needed. This causes unneeded calls to ast_optional_api_use. * Don't include optional_api.h from sip_api.h. * Move 'struct ast_channel_monitor' to channel.h. * Don't include monitor.h from chan_sip.c, channel.c or features.c. The move of struct ast_channel_monitor is needed since channel.c depends on it. This has no effect on users of monitor.h since channel.h is included from monitor.h. ASTERISK-25051 #close Reported by: Corey Farrell Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478 |
11 years ago |
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32eb00b70a |
Merge "Astobj2: Fix initialization order of refdebug and AO2_DEBUG."
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11 years ago |
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e4f0a55f7f |
term: send proper reset sequence when black background is forced
When using the force black background command-line option or configuration option an invalid reset sequence is sent following a coloured output item in the CLI, the result is that the colour is not 'turned off' and continues until the next non-default coloured text output. A reset sequence is already defined in term.c, but the ast_term_reset function doesn't use it, instead building it's own invalid sequence and returning that. This patch changes that behaviour, removing the building of a reset sequence and instead using the pre-built constant 'enddata' which is a suitable reset sequence for this purpose. ASTERISK-24896 #close Reported by: Dan Tucny Change-Id: I56323899123ae3264900389cae1f5b252aa3bf43 |
11 years ago |
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8f3cee1258 |
Astobj2: Fix initialization order of refdebug and AO2_DEBUG.
This ensures that refdebug is initialized before AO2_DEBUG if both are enabled, since AO2_DEBUG allocates a container. This change also makes AO2_DEBUG initialization critical, a failure will abort Asterisk startup. This is needed since the failure would be caused by reg_containers allocation failure, and that would result in a segmentation fault by ao2_container_register later in startup. ASTERISK-25048 #close Reported by: Corey Farrell Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244 |
11 years ago |
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7ac28be04b |
main/pbx: Improve performance of dialplan reloads with a large number of hints
The PBX core maintains two hash tables for hints: a container of the actual hints (hints), along with a container of devices that are watching that hint (hintdevices). When a dialplan reload occurs, each hint in the hints container is destroyed; this requires a lookup in the container of devices to find the device => hint mapping object. In the current code, this performs an ao2_callback, iterating over each of the device to hint objects in the hintdevices container. For a large number of hints, this is extremely expensive: dialplan reloads with 20000 hints could take several minutes in just this phase. This patch improves the performance of this step in the dialplan reloads by caching which devices are watching a hint on the hint object itself. Since we don't want to create a circular reference, we just cache the name of the device. This allows us to perform a smarter ao2_callback on the hintdevices container during hint removal, hashing on the name of the device and returning an iterator to the matching names. The overall performance improvement is rather large, taking this step down to a number of seconds as opposed to minutes. In addition, this patch also registers the hint containers in the PBX core with the astobj2 library. This allows for reasonable debugging to hash collisions in those containers. ASTERISK-25040 #close Reported by: Matt Jordan Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360 |
11 years ago |
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dc23204aca |
Prevent potential crash on blond transfer.
Scenario: Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects the incoming call (or some other immediate circumstance causes Carol not to answer the call) What occurs in this case is that when the bridge between Alice and Bob breaks, Alice is told to masquerade into Bob's channel that had placed the call to Carol. The actual masquerade goes down without a hitch. However, a channel fixup callback that attempts to publish dial events over Stasis has a crash. The reason for this crash is that the datastore on Bob's channel that placed the outbound call to Carol only had a bare pointer to Carol's channel. Since Carol rejected the incoming call, Carol's channel has been hung up and freed, meaning accessing her channel results in a crash. The fix here is simple. The dial fixup code has been altered to hold references to the involved channels and to drop those references when freeing data. ASTERISK-25025 #close Reported by Chet Stevens Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197 |
11 years ago |
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11ffcf662f |
Restrict functionality when ACLs are misconfigured.
This patch has two main purposes: 1) Improve warning messages when ACLs are configured improperly. 2) Prevent misconfigured ACLs from allowing potentially unwanted traffic. To acomplish point (2) in most cases, whatever configuration object that the ACL belonged to was not allowed to load. The one exception is res_pjsip_acl. In that case, ACLs are their own configuration object. Furthermore, the module loading code has no indication that a ACL configuration had a failure. So the tactic taken here is to create an ACL that just blocks everything. ASTERISK-24969 Reported by Corey Farrell Change-Id: I2ebcb6959cefad03cea4d81401be946203fcacae |
11 years ago |
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39d3e1ef6e |
main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8
The patch in
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11 years ago |
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57cbb4bc8d |
Merge "Astobj2: Add ao2_weakproxy_ref_object function."
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11 years ago |
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c9c03998cc |
Astobj2: Add ao2_weakproxy_ref_object function.
This function allows code to run ao2_ref against the real object associated with a weakproxy. It is useful when all of the following conditions are true: * You have a pointer to weakproxy. * You do not have or need a pointer to the real object. * You need to ensure the real object exists and is not destroyed during a process. In this case it's wasteful to store a pointer to the real object just for the sake of releasing it later. Change-Id: I38a319b83314de75be74207a8771aab269bcca46 |
11 years ago |
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55a780d211 |
Git Conversion: Switch Non-C files to ASTERISK_REGISTER_FILE.
This switches files used to generate other sources to use the new ASTERISK_REGISTER_FILE macro. ASTERISK-25026 #close Reported by: Corey Farrell Change-Id: Ieb2537b83421cad07c8955e5f90c405ccf079740 |
11 years ago |
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5e96584829 |
res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS
ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created. The resources are linked into a table, but the original alloc refs are never released. ast_strdup leak in rtp_engine.c. If ast_rtp_dtls_cfg_copy() is called twice on the same destination struct, a pointer to an alloc'd string is overwritten before the string is free'd. ASTERISK-25022 Reported by: one47 Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b |
11 years ago |
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5c1d07baf0 |
Astobj2: Allow reference debugging to be enabled/disabled by config.
* The REF_DEBUG compiler flag no longer has any effect on code that uses Astobj2. It is used to determine if reference debugging is enabled by default. Reference debugging can be enabled or disabled in asterisk.conf. * Caller information is provided in logger errors for ao2 bad magic numbers. * Optimizes AO2 by merging internal functions with the public counterpart. This was possible now that we no longer require a dual ABI. ASTERISK-24974 #close Reported by: Corey Farrell Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1 |
11 years ago |
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ca7193167e |
Clang: change previous tautological-compare fixes.
clang can warn about a so called tautological-compare, when it finds comparisons which are logically always true, and are therefor deemed unnecessary. Exanple: unsigned int x = 4; if (x > 0) // x is always going to be bigger than 0 Enum Case: Each enumeration is its own type. Enums are an integer type but they do not have to be *signed*. C leaves it up to the compiler as an implementation option what to consider the integer type of a particu- lar enumeration is. Gcc treats an enum without negative values as an int while clang treats this enum as an unsigned int. rmudgett & mmichelson: cast the enum to (unsigned int) in assert. The cast does have an effect. For gcc, which seems to treat all enums as int, the cast to unsigned int will eliminate the possibility of negative values being allowed. For clang, which seems to treat enums without any negative members as unsigned int, the cast will have no effect. If for some reason in the future a negative value is ever added to the enum the assert will still catch the negative value. ASTERISK-24917 Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a |
11 years ago |
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7ccaf8aa46 |
Merge "Astobj2: Ensure all calls to __adjust_lock pass a valid object."
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11 years ago |
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5ae61cf1eb |
Merge "New AMI Command Output Format"
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11 years ago |
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c231c85ea4 |
Astobj2: Ensure all calls to __adjust_lock pass a valid object.
__adjust_lock doesn't check for invalid objects, and doesn't have an appropriate return value for invalid objects. Most callers of __adjust_lock pass objects that have already been confirmed valid, this change adds checks before the remaining calls. ASTERISK-24997 #close Reported by: Corey Farrell Change-Id: I669100f87937cc3f867cec56a27ae9c01292908f |
11 years ago |
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7216e3c608 |
dns: Make query sets hold on to queries for their lifetime.
The query set documentation states that upon completion queries can be retrieved for the lifetime of the query set. This is a reasonable expectation but does not currently occur. This was originally done to resolve a circular reference between queries and query sets, but in practice the query can be kept. This change makes it so a query does not have a reference to the query set until it begins resolving. It also makes it so that the reference is given up upon the query being completed. This allows the queries to remain for the lifetime of the query set. As the query set on the query is only useful to the query set functionality and only for the lifetime that the query is resolving this is safe to do. ASTERISK-24994 #close Reported by: Joshua Colp Change-Id: I54e09c0cb45475896654e7835394524e816d1aa0 |
11 years ago |
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5757d2d30d |
Check for ao2_alloc failure in __ast_channel_internal_alloc.
Fix a crash that could occur in __ast_channel_internal_alloc if ao2_alloc fails. ASTERISK-24991 #close Change-Id: I4ca89189eb22f907408cb87d0a1645cfe1314a90 |
11 years ago |
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2f418c052e |
New AMI Command Output Format
This change modifies how the the output from a CLI command is sent to a client over AMI. Output from the CLI command is now sent as a series of zero-or-more Output: headers. Additionally, commands that fail to execute (eg: no such command, invalid syntax etc.) now cause an Error response instead of Success. If the command executed successfully, but the manager unable to provide the output the reason will be included in the Message: header. Otherwise it will contain 'Command output follows'. Depends on a new version of starpy (> 1.0.2) that supports the new output format. See pull-request https://github.com/asterisk/starpy/pull/34 ASTERISK-24730 Change-Id: I6718d95490f0a6b3f171c1a5cdad9207f9a44888 |
11 years ago |
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8e903b17ea |
main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple
When a PBX registrar is unloaded, it will fail to remove its extension from the context root_table if a dialplan application used by that extension is still loaded. This can be the case for AGI, which can be unloaded after several of the standard PBX providers. Often, this is harmless; however, if the extension's priorities are removed during the failed unloading *and* the dialplan application later unregisters, it leaves a ticking timebomb for the next PBX provider that attempts to iterate over the extensions. When that occurs, the peer_table pointer on the extension will already be set to NULL. The current code does not check to see if the pointer is NULL before passing it to a hashtab function this is not NULL tolerant. Since it is possible for the peer_table to be NULL when we normally would not expect that to be the case, the solution in this patch is to simply skip over processing an extension's priorities if peer_table is NULL. Prior to this patch, the tests/pbx/callerid_match test would crash during module unload. With this patch, the test no longer crashes after running. ASTERISK-24774 #close Reported by: Corey Farrell Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40 |
11 years ago |
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8435a0cdff |
Merge "Detect potential forwarding loops based on count."
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11 years ago |
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aae45acbda |
Detect potential forwarding loops based on count.
A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23 |
11 years ago |
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bb347fa594 |
Merge topic 'ASTERISK-24863'
* changes: res_pjsip: Add global option to limit the maximum time for initial qualifies pjsip_options: Add qualify_timeout processing and eventing res_pjsip: Refactor endpt_send_request to include transaction timeout |
11 years ago |
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56a2baa21d |
bridge.c: NULL app causes crash during attended transfer
Due to a race condition there was a chance that during an attended transfer the channel's application would return NULL. This, of course, would cause a crash when attempting to access the memory. This patch retrieves the channel's app at an earlier time in processing in hopes that the app name is available. However, if it is not then "unknown" is used instead. Since some string value is now always present the crash can no longer occur. ASTERISK-24869 #close Reported by: viniciusfontes Review: https://gerrit.asterisk.org/#/c/133/ Change-Id: I5134b84c4524906d8148817719d76ffb306488ac |
11 years ago |
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51886c68dc |
pjsip_options: Add qualify_timeout processing and eventing
This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com> |
11 years ago |
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a3cec44a0a |
res_pjsip: Add external PJSIP resolver implementation using core DNS API.
This change adds the following: 1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked. 2. Unit tests for the query set implementation. 3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups. For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A, with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit transport has been provided. Configured transports on the system are taken into account to eliminate resolved addresses which have no hope of completing. ASTERISK-24947 #close Reported by: Joshua Colp Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e |
11 years ago |
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7687188a2d |
Merge "astobj2: Add support for weakproxy objects."
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11 years ago |
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62e95065d6 |
AMI: Fix improper handling of lines that are exactly 1025 bytes long.
When AMI receives a line that is 1025 bytes long, it sends two error messages. Copy the last byte in the buffer to the first postiion, set the length to 1. ASTERISK-20524 #close Reported by: David M. Lee Change-Id: Ifda403e2713b59582c715229814fd64a0733c5ea |
11 years ago |
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cb6bf3094e |
astobj2: Add support for weakproxy objects.
This implements "weak" references. The weakproxy object is a real ao2 with normal reference counting of its own. When a weakproxy is pointed to a normal object they hold references to each other. The normal object is automatically freed when a single reference remains (the weakproxy). The weakproxy also supports subscriptions that will notify callbacks when it does not point to any real object. ASTERISK-24936 #close Reported by: Corey Farrell Change-Id: Ib9f73c02262488d314d9d9d62f58165b9ec43c67 |
11 years ago |