- Mar 20, 2017
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Aaron An authored
Fixed a bug in function "ast_audiohook_write_frame" that checked the variable other_factory_samples and only flushed the factories, so they would be in sync, when other_factory_samples > 0. When there is not any rtp incoming the variable other_factory_samples will be 0, and although the result of "our_factory_ms - other_factory_ms" may be very large, this led to the record file not syncing. ASTERISK-26875 #close Reported-by: Aaron An Tested-by: Aaron An Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22
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- Mar 15, 2017
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zuul authored
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- Mar 14, 2017
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zuul authored
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Joshua Colp authored
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- Mar 13, 2017
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Joshua Colp authored
When querying for PJSIP specific information using the dialplan function CHANNEL() it is possible that the underlying session will no longer have a channel associated with it. This is most likely to occur when the RTCP HEP module attempts to get the channel name. If this happens then a crash will occur. This change just adds a check that the channel exists on the session before querying it. ASTERISK-26857 Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01
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- Mar 11, 2017
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George Joseph authored
Bundled pjproject should now only rebuild if one of the menuselect "Compiler Flags" options changes. Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43
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- Mar 10, 2017
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Joshua Colp authored
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zuul authored
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- Mar 09, 2017
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Joshua Colp authored
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Daniel Journo authored
* cli_commands.c Fixed CLI output ASTERISK-26822 #close Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01
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Joshua Colp authored
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Daniel Journo authored
* res_musiconhold.c: Ensure the general section is not treated as a moh class. ASTERISK-26353 #close Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d
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- Mar 08, 2017
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Sean Bright authored
Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def
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Sean Bright authored
Set a variable on the channel that indicates which attempt number we are currently performing to allow for attempt-specific behavior. ASTERISK-26568 #close Reported by: Roman Shubovich Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89
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Joshua Colp authored
This change adds a PJSIP patch (which has been contributed upstream) to allow the registration of IPv6 transport types. Using this the res_pjsip_transport_websocket module now registers an IPv6 Websocket transport and uses it for the corresponding traffic. ASTERISK-26685 Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647
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Daniel Journo authored
* apps/app_voicemail.c fromstring field added to mailbox which will override the global fromstring if set. ASTERISK-24562 #close Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe
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zuul authored
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- Mar 07, 2017
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zuul authored
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Mark Michelson authored
When doing some WebRTC testing, I found that the websocket would disconnect whenever I attempted to place a call into Asterisk. After looking into it, I pinpointed the problem to be due to the iostreams change being merged in. Under certain circumstances, a call to ast_iostream_read() can return a negative value. However, in this circumstance, the websocket code was treating this negative return as if it were a partial read from the websocket. The expected length would get adjusted by this negative value, resulting in the expected length being too large. This patch simply adds an if check to be sure that we are only updating the expected length of a read when the return from a read is positive. ASTERISK-26842 #close Reported by Mark Michelson Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab
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Jean Aunis authored
When receiving a 422 response, the invitestate variable must be reset to INV_CALLING. ASTERISK-26841 Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099
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- Mar 06, 2017
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Daniel Journo authored
* say.c Changed 'digits/and' to 'vm-and' for en_GB ASTERISK-26598 #close Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe
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Sean Bright authored
Per the linked issue, we aren't checking the buffer filled by fgets() to determine if it contains a newline, so we will fail to correctly parse the trailing portion of a long line. This patch increases the buffer size from 256 to 1024, and skips any line that exceeds that length, logging a warning in the process. ASTERISK-17067 #close Reported by: Dave Olszewski Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0
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- Mar 03, 2017
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Richard Mudgett authored
* manager.c:manager_state_cb() Fix potential use of uninitialized hint[] if a hint does not exist for the requested extension. Ran into this when developing a testsuite test. The AMI event ExtensionStatus came out with the hint header value containing garbage. The AMI event PresenceStatus also had the same issue. * manager.c:action_extensionstate() no need to completely initialize the hint[]. Only initialize the first element. * pbx.c:ast_add_hint() Remove unnecessary assignment. * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care about the return value of ast_get_hint() there. Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b
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- Mar 01, 2017
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Joshua Colp authored
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Joshua Colp authored
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Joshua Colp authored
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Jørgen H authored
According to the RFC[1] WSS should only be used in the Via header for secure Websockets. * Use WSS in Via for secure transport. * Only register one transport with the WS name because it would be ambiguous. Outgoing requests may try to find the transport by name and pjproject only finds the first one registered. This may mess up unsecure websockets but the impact should be minimal. Firefox and Chrome do not support anything other than secure websockets anymore. * Added and updated some debug messages concerning websockets. * security_events.c: Relax case restriction when determining security transport type. * The res_pjsip_nat module has been updated to not touch the transport on Websocket originating messages. [1] https://tools.ietf.org/html/rfc7118 ASTERISK-26796 #close Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12
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George Joseph authored
* Removed the AST_CHAN_TP_MULTISTREAM tech property. We now rely on read_stream being set to indicate a multi stream channel. * Added ast_channel_is_multistream convenience function. * Fixed issue where stream and default_stream weren't being set on a frame retrieved from the queue. * Now testing for NULL being returned from the driver's read or read_stream callback. * Fixed issue where the dropnondefault code was crashing on a NULL f. * Now enforcing that if either read_stream or write_stream are set when ast_channel_tech_set is called that BOTH are set. * Added the unit tests. ASTERISK-26816 Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2
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Sean Bright authored
res_config_pgsql should match the behavior of other realtime backend drivers so that queue_log can disable adaptive logging. ASTERISK-25628 #close Reported by: Dmitry Wagin Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372
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Mark Michelson authored
This introduces and documents the various states in the state machine. This also introduces API functions that induce state changes, and places TODO comments telling what needs to be done in addition to what is already there. Those TODOs will be replaced with real code in upcoming changes. Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed
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Joshua Colp authored
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Joshua Colp authored
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Joshua Colp authored
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- Feb 28, 2017
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Joshua Colp authored
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Joshua Colp authored
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zuul authored
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Sean Bright authored
In the event that a cache file is removed out from under us, we should treat the cache entry as stale and force a refresh. ASTERISK-26774 #close Reported by: Igor Gamayunov Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52
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Joshua Colp authored
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Joshua Colp authored
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Sean Bright authored
The find_table() functions NULL or a locked table pointer. We are not consistently calling release_table() in failure paths. Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544
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