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${ noResults }
27222 Commits (0405c31756e45047cec20caccf228e5862550f9c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
e0090216db |
CLI: Cosmetic issue - core show uptime
Show uptime information ends with an unnecessary space. Now NEEDCOMMA is better defined. Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1 |
10 years ago |
|
88212ccb7f |
res_pjsip: Prevent access of NULL channels.
It is possible to receive incoming requests or responses after the channel on an ast_sip_session has been destroyed and NULLed out. Handlers of these sorts of requests or responses need to be prepared for the possibility that the channel is NULL or else they could cause a crash. While several places have been amended to deal with NULL channels, there were still a couple of places that needed updating. res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to return early if there is no channel on the session. res_pjsip_session.c: When handling a 302 response, we need to stop the redirecting attempt if there is no channel on the session. ASTERISK-25148 #close reported by Mark Michelson Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9 |
10 years ago |
|
f5d5aa67dc |
AMI: Escape string values.
So this issue is a bit complicated. Since it is possible to pass values to AMI that contain a '\r\n' (or other similar sequences) these values need to be escaped. One way to solve this is to escape the values and then pass the escaped values to the AMI variable parameter string building function. However, this puts the onus on the pre-build function to escape all string values. This potentially requires a fair amount of changes along with a lot of string allocations/freeing for all values. Surely there is a way to push this complexity down a level into the string building function itself? This of course is possible, but ends up requiring a way to distinguish between strings that need to be escaped and those that don't. The best way to handle this is by introducing a new format specifier in the format string. For instance a %s (no escape) and %S (escape). However, that is a bit weird and unexpected. So faced with those possibilities this patch implements a limited version of the first option. Instead of attempting to escape all string values this patch only escapes those values that make sense. This approach limits the number of changes and doesn't suffer from the odd format specifier problem. ASTERISK-24934 #close Reported by: warren smith Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0 |
10 years ago |
|
5dc9fb4198 |
res_pjsip/location: Fix ref leak in contact_apply_handler
contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status to force the creation of a contact_status object whenever a new contact is added but it didn't unref the returned object. Added an ao2_cleanup(status) to plug the leak. ASTERISK-25141 Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40 Reported-by: Corey Farrell |
10 years ago |
|
d908272b7e |
Fixes for OS X
* Add some type casting so tv_usec can really be a long, instead of some strange platform specific type. * Add some .dylib style files to .gitignore. * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer versions of GCC, when compiling the Homebrew formula for Asterisk, are not properly passing the -Xlinker options to the linker. Given that -Wl, does exactly the [same thing][], and does it properly, this patch changes the -Xlinker options to use -Wl, instead. [reasons unknown]: http://bit.ly/1SUbEYx [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd |
10 years ago |
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bc70904c05 |
Merge "res_pjsip_session: Fix in-dialog authentication." into 13
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10 years ago |
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1e3701a529 |
Merge "Fix buffer overflow in slin sample frames generation." into 13
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10 years ago |
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9e7827e3ac |
pjsip_configuration: Fix leak in persistent_endpoint_update_state.
The loop to find the first available contact of an endpoint grabbed contact from the iterator, then checked for offline state. This caused the first contact after the state was found to leak a reference. ASTERISK-25141 Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08 |
10 years ago |
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888bb49618 |
Fix buffer overflow in slin sample frames generation.
The length of frames retured by sample functions was twice as large as real, what caused global buffer overflow caught by AddressSanitizer. ASTERISK-24717 #close Reported by: Badalian Vyacheslav Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6 |
10 years ago |
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857166b5e5 |
res_pjsip/location: Fix memory leak in permanent_uri_handler
When permanent_uri_handler was creating the contact status object for each contact, it wasn't unreffing it at the end of the loop. ASTERISK-25141 #close Reported-by: Corey Farrell Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12 |
10 years ago |
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1558a89129 |
Revert "endpoint/stasis: Eliminate duplicate events on endpoint status change"
This reverts commit
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10 years ago |
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35c699086a |
endpoint/stasis: Eliminate duplicate events on endpoint status change
When an endpoint was created, it's messages were being forwarded to both the tech endpoint topic and the all endpoints topic. Since the tech topic was also forwarded to all, this was resulting in duplicate messages whenever an endpoint published. This patch causes the endpoint to only forward to the tech topic and lets the tech topic forward to all. To accomplish this, the existing stasis_cp_single_create function (which both creates and forwards) was cloned and split into 2 functions, one that creates the topic and one that sets up the forwarding. This allows endpoint_internal_create to create the topic from the endpoint_all cache without forwarding it there, then allows it to do the forward to the tech's topic. ASTERISK-25137 #close Reported-by: Vitezslav Novy ASTERISK-25116 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c |
10 years ago |
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fe21f2e52f |
res_pjsip_session: Fix in-dialog authentication.
When the remote peer requires authentication for in-dialog requests then re-INVITEs to the peer cause the call to be disconnected and other in-dialog requests to the peer like MESSAGE just don't go through. * Made session_inv_on_tsx_state_changed() handle in-dialog authentication for re-INVITEs and other methods. Initial INVITEs cannot be handled here because the INVITE transaction must be restarted earlier. * Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in preparation for removing the file. The generic outbound authentication code did not work as well as anticipated. * Created outbound_invite_auth() to only handle initial outbound INVITEs. Re-INVITEs cannot be handled here. The re-INVITE transaction is still in progress and the PJSIP library cannot handle the overlapping INVITE transactions. Other method types should not be handled here as this code only works on outgoing calls and we need to handle incoming and outgoing calls. ASTERISK-25131 #close Reported by: Richard Mudgett Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0 |
10 years ago |
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262d590819 |
res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes
Add a new ContactStatus AMI event. Publish the following status/state changes: Created Removed Reachable Unreachable Unknown Contact URI, new status/state, aor and endpoint names, and the last qualify rtt result are included in the event. ASTERISK-25114 #close Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> |
10 years ago |
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87c03b792b |
Merge "Astobj2: Correctly treat hash_fn returning INT_MIN" into 13
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10 years ago |
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5a42397018 |
sorcery: Fix cache creation callback.
The cache creation callback function expects to receive a sorcery_details structure and not just a standalone object. Change-Id: I3e4a5a137cb25292eb52d7a14cbb6daa09213450 |
10 years ago |
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97a6ce1717 |
Astobj2: Correctly treat hash_fn returning INT_MIN
The code in astobj2_hash.c wrongly assumed that abs(int) is always > 0. However, abs(INT_MIN) = INT_MIN and is still negative, as well as abs(INT_MIN) % num_buckets, and as a result this led to a crash. One way to trigger the bug is using host=::80 or 0.0.0.128 in peer configuration section in chan_sip or chan_iax. This patch takes the remainder before applying abs, so that bucket number is always in range. ASTERISK-25100 #close Reported by: Mark Petersen Change-Id: Id6981400ad526f47e10bcf7b847b62bd2785e899 |
10 years ago |
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b9826bf101 |
Merge "Stasis: Fix unsafe use of stasis_unsubscribe in modules." into 13
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10 years ago |
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554bd1e39c |
res_pjsip_transport_websocket: Fix crash on receiving large SIP packets
Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves truncated before passing to pjsip_tpmgr_receive_packet, but the length was passed unaltered, thus causing memory corruption and segfault. ASTERISK-25122 #close Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab |
10 years ago |
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0d266cbe02 |
Stasis: Fix unsafe use of stasis_unsubscribe in modules.
Many uses of stasis_unsubscribe in modules can be reached through unload. These have been switched to stasis_unsubscribe_and_join. Some subscription callbacks do nothing, for these I've created a noop callback function in stasis.c. This is used by some modules that monitor MWI topics in order to enable cache, since the callback does not become invalid after dlclose it is safe to use stasis_unsubscribe on these, even during module unload. ASTERISK-25121 #close Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c |
10 years ago |
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eaabc4d04c |
Merge "res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS" into 13
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10 years ago |
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4690bc81f7 |
Merge "res/res_pjsip_exten_state: Fix confusing NOTICE message" into 13
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10 years ago |
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51ffed5e61 |
res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS
In addition to specifying lists of 'presence' and 'message-summary', users can also create lists of type 'dialog'. These should be treated in the same fashion as 'presence'. Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e |
10 years ago |
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7950b65e4f |
res/res_pjsip_exten_state: Fix confusing NOTICE message
When a SUBSCRIBE request is made to a dialplan hint that doesn't exist, the current NOTICE message informing users of this swaps the context and extension parameters. This can cause a bit of confusion. Thanks to CptBurger in #asterisk for helping to point this out. Change-Id: Ie584d1a58ae217385c87a450ca25b55ca0e36e43 |
10 years ago |
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ea8620a51b |
Merge "res/ari: Register Stasis application on WebSocket attempt" into 13
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10 years ago |
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5ac65ddfb4 |
res/ari: Register Stasis application on WebSocket attempt
Prior to this patch, when a WebSocket connection is made, ARI would not be informed of the connection until after the WebSocket layer had accepted the connection. This created a brief race condition where the ARI client would be notified that it was connected, a channel would be sent into the Stasis dialplan application, but ARI would not yet have registered the Stasis application presented in the HTTP request that established the WebSocket. This patch resolves this issue by doing the following: * When a WebSocket attempt is made, a callback is made into the ARI application layer, which verifies and registers the apps presented in the HTTP request. Because we do not yet have a WebSocket, we cannot have an event session for the corresponding applications. Some defensive checks were thus added to make the application objects tolerant to a NULL event session. * When a WebSocket connection is made, the registered application is updated with the newly created event session that wraps the WebSocket connection. ASTERISK-24988 #close Reported by: Joshua Colp Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636 |
10 years ago |
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1b475a8410 |
Merge "res_pjsip: Refactor endpt_send_transaction (qualify_timeout)" into 13
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10 years ago |
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02dfb118ba |
Merge "res_pjsip_outbound_registration: Check request URI for line." into 13
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10 years ago |
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60e2fbfe62 |
res_pjsip: Refactor endpt_send_transaction (qualify_timeout)
This patch refactors the transaction timeout processing to eliminate calling the lower level public pjsip functions and reverts to calling pjsip_endpt_send_request again. This is the result of me noticing a possible incompatibility with pjproject-2.4 which was causing contact status flapping. The original version of this feature used the lower level calls to get access to the tsx structure in order to cancel the transaction when our own timer expires. Since we no longer have that access, if our own timer expires before the pjsip timer, we call the callbacks and just let the pjsip transaction take it's own course. When the transaction ends, it discovers the callbacks have already been run and just cleans itself up. A few messages in pjsip_configuration were also added/cleaned up. ASTERISK-25105 #close Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> |
10 years ago |
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42476e6633 |
res_pjsip_outbound_registration: Check request URI for line.
When an inbound call is received the To header is checked for the "line" option. Some remote servers will place this in the request URI instead. This adds an additional check for the option in the request URI. ASTERISK-25072 #close Reported by: Dmitriy Serov Change-Id: Id4e44debbb80baad623b914a88574371575353c8 |
10 years ago |
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e7edb59db6 |
res_mwi_external_ami: Use module version of AMI registration.
Use ast_manager_register_xml for res_mwi_external_ami manager actions. This ensures the module is held open while any of the actions are being run. ASTERISK-25117 #close Reported by: Corey Farrell Change-Id: Iececfdc2da498b2c32b9e09042f5f12292007ac7 |
10 years ago |
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9d8a462356 |
ARI: Update version to 1.7.0
This patch updates the version of ARI to 1.7.0 to reflect the backwards compatible changes that will be introduced in 13.4.0. Change-Id: I6c36e6144da426412f25828a868e4df916bff60a |
10 years ago |
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620054c527 |
Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets" into 13
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10 years ago |
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f5e195b44e |
Merge "Logger: Reset defaults before processing config." into 13
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10 years ago |
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e8a4e01c32 |
Merge "res/res_http_websocket: Add a pre-session established callback" into 13
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10 years ago |
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3c98544543 |
Merge "main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits" into 13
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10 years ago |
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9b6e228419 |
Logger: Reset defaults before processing config.
Reset options to default values before reloading config. This ensures that if a setting is removed or commented out of the configuration file it is unset on reload. ASTERISK-25112 #close Reported by: Corey Farrell Change-Id: Id24bb1fb0885c2c14cf8bd6f69a0c2ee7cd6c5bd |
10 years ago |
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7fcf0a97b8 |
app_playback: Suppress warnings on playback if channel hung up
If a channel hangs up while an audio file is playing, there's no need to clutter up the logs with a warning so suppress it if ast_check_hangup returns true. Also, change warning to debug/2 in file.c if writing a frame fails. Same reasoning. Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89 Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> |
10 years ago |
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b1e8c0b9eb |
audiohook.c: Difference in read/write rates caused continuous buffer resets
Currently, everytime a sample rate change occurs (on read or write) the associated factory buffers are reset. If the requested sample rate on a read differed from that of a write then the buffers are continually reset on every read and write. This has the side effect of emptying the buffer, thus there being no data to read and then write to a file in the case of call recording. This patch fixes it so that an audiohook_list's rate always maintains the maximum sample rate among hooks and formats. Audiohook sample rates are only overwritten by this value when slin native compatibility is turned on. Also, the audiohook sample rate can only overwrite the list's sample rate when its rate is greater than that of the list or if compatibility is turned off. This keeps the rate from constantly switching/resetting. ASTERISK-24944 #close Reported by: Ronald Raikes Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f |
10 years ago |
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4a450f863b |
Merge "Fix potential crash after unload of func_periodic_hook or test_message." into 13
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10 years ago |
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17d6ede337 |
main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits
ASTERISK-24887 #close Reported by: Makoto Dei Tested by: tensai Change-Id: I6a96f572adb17f76b3acafe503a01c48eb5dd9bf |
10 years ago |
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31cc24aad6 |
res/res_http_websocket: Add a pre-session established callback
This patch updates http_websocket and its corresponding implementation with a pre-session established callback. This callback allows for WebSocket server consumers to be notified when a WebSocket connection is attempted, but before we accept it. Consumers can choose to reject the connection, if their application specific logic allows for it. As a result, this patch pulls out the previously private websocket_protocol struct and makes it public, as ast_websocket_protocol. In order to preserve backwards compatibility with existing modules, the existing APIs were left as-is, and new APIs were added for the creation of the ast_websocket_protocol as well as for adding a sub-protocol to a WebSocket server. In particular, the following new API calls were added: * ast_websocket_add_protocol2 - add a protocol to the core WebSocket server * ast_websocket_server_add_protocol2 - add a protocol to a specific WebSocket server * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object. Consumers can populate this with whatever callbacks they wish to support, then add it to the core server or a specified server. ASTERISK-24988 Reported by: Joshua Colp Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2 |
10 years ago |
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f9114179e6 |
chan_pjsip: Fix crash during off-nominal when no endpoint specified.
Add missing return -1 when no endpoint name is specified. ASTERISK-25086 #close Reported by: snuffy Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e |
10 years ago |
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dd78ab42e4 |
res_pjsip_config_wizard/config: Fix template processing
The config wizard was always pulling the first occurrence of a variable from an ast_variable list but this gets the template value from the list instead of any overridden value. This patch creates ast_variable_find_last_in_list() in config.c and updates res_pjsip_config_wizard to use it instead of ast_variable_find_in_list. Now the overridden values, where they exist, are used instead of template variables. Updated test_config to test the new API. ASTERISK-25089 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4 |
10 years ago |
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091b436007 |
cdr: Fix 'core show channel' CDR variable truncation.
When the new Bridging API was implemented, the workspace variable changed to a malloc'd string, causing sizeof() to always be 8 (char). Revert back to stored on stack string for workspace. ASTERISK-25090 #close Change-Id: I51e610ae87371df771ce7693a955510efb90f8f7 |
10 years ago |
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8697a49ef9 |
Merge "sorcery: Add API to insert/remove a wizard to/from an object type's list" into 13
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10 years ago |
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aea349a87e |
Merge "Message.c: Clear message channel frames on cleanup" into 13
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10 years ago |
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6b7282ca40 |
Fix potential crash after unload of func_periodic_hook or test_message.
These modules save a pointer to the context they create on load, and use that pointer to destroy the context at unload. It is not safe to save this pointer, it is replaced during load of pbx_config, pbx_lua or pbx_ael. This change causes the modules to pass NULL to ast_context_destroy, a safer way to perform the unregistration since it does not use a pointer that could become invalid. ASTERISK-25085 #close Reported by: Corey Farrell Change-Id: I6a00ec8e38046058f97dc703e1adcde9bf517835 |
10 years ago |
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8f8d54a18e |
Merge "main/manager.c: Bugfix sort action_manager by alphabetically" into 13
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10 years ago |
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02c5130589 |
Message.c: Clear message channel frames on cleanup
The message channel is a special channel that doesn't actually process frames. However, certain actions can cause frames to be placed in the channel's read queue including the Hangup application which is called on the channel after each message is processed. Since the channel will continually be reused for many messages, it's necessary to flush these frames at some point. ASTERISK-25083 #close Reported by: Jonathan Rose Change-Id: Idf18df73ccd8c220be38743335b5c79c2a4c0d0f |
10 years ago |