4943 Commits (39cc28f6ea2140ad6d561fd4c9e9a66f065cecee)

Author	SHA1	Message	Date
Joshua Colp	20ee33e22e	Merge topic 'misc_rtp_tweaks' into 13 ... * changes: rtp_engine.h: No sense allowing payload types larger than RFC allows. rtp_engine.c: Minor tweaks. rtp_engine.h: Misc comment fixes. chan_sip.c: Tweak glue->update_peer() parameter nil value.	11 years ago
Mark Michelson	e28fbebc57	Merge "ARI: Rotate log channels." into 13	11 years ago
Benjamin Ford	1ae762634c	ARI: Rotate log channels. ... An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01	11 years ago
Richard Mudgett	89b21fd9a3	rtp_engine.h: No sense allowing payload types larger than RFC allows. ... * Tweaked add_static_payload() to not use magic numbers. Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b	11 years ago
Richard Mudgett	7427c7f13b	rtp_engine.c: Minor tweaks. ... * Fix off nominial ref leak of new_type in ast_rtp_codecs_payloads_set_m_type(). * No need to lock static_RTP_PT_lock in ast_rtp_codecs_payloads_set_m_type() and ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type parameter sanity check. * No need to create ast_rtp_payload_type ao2 objects with a lock since the lock is not used. Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4	11 years ago
Mark Michelson	10ba72a927	Add a test event for inband ringing. ... This event is necessary for the bridge_wait_e_options test to be able to confirm that ringing is being played on the local channel that runs the BridgeWait() application with the e(r) option. ASTERISK-25292 #close Reported by Kevin Harwell Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e	11 years ago
Mark Michelson	c9099d06cc	Merge "holding_bridge: ensure moh participants get frames" into 13	11 years ago
Jonathan Rose	8458b8d441	holding_bridge: ensure moh participants get frames ... Currently, if a blank musiconhold.conf is used, musiconhold will fail to start for a channel going into a holding bridge with an anticipation of getting music on hold. That being the case, no frames will be written to the channel and that can pose a problem for blind transfers in PJSIP which may rely on frames being written to get past the REFER framehook. This patch makes holding bridges start a silence generator if starting music on hold fails and makes it so that if no music on hold functions are installed that the ast_moh_start function will report a failure so that consumers of that function will be able to respond appropriately. ASTERISK-25271 #close Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99	11 years ago
Joshua Colp	2749721791	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	11 years ago
Mark Michelson	f635520527	Local channels: Alternate solution to ringback problem. ... Commit 54b25c80c8 solved an issue where a specific scenario involving local channels and a native local RTP bridge could result in ringback still being heard on a calling channel even after the call is bridged. That commit caused many tests in the testsuite to fail with alarming consequences, such as not sending DialBegin and DialEnd events, and giving incorrect hangup causes during calls. This commit reverts the previous commit and implements and alternate solution. This new solution involves only passing AST_CONTROL_RINGING frames across local channels if the local channel is in AST_STATE_RING. Otherwise, the frame does not traverse the local channels. By doing this, we can ensure that a playtones generator does not get started on the calling channel but rather is started on the local channel on which the ringing frame was initially indicated. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39	11 years ago
Matt Jordan	4d8f47f4bf	Merge "audiohook: Use manipulated frame instead of dropping it." into 13	11 years ago
Joshua Colp	ff83c115c7	Merge "Local channels: Do not block control -1 payloads." into 13	11 years ago
Joshua Colp	f509730cb9	audiohook: Use manipulated frame instead of dropping it. ... Previous changes to sample rate support in audiohooks accidentally removed code responsible for allowing the manipulate audiohooks to work. Without this code the manipulated frame would be dropped and not used. This change restores it. ASTERISK-25253 #close Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13	11 years ago
Mark Michelson	54b25c80c8	Local channels: Do not block control -1 payloads. ... Control frames with a -1 payload are used as a special signal to stop playtones generators on channels. This indication is sent both by app_dial as well as by ast_answer() when a call is answered in case any tones were being generated on a calling channel. This control frame type was made to stop traversing local channel pairs as an optimization, because it was thought that it was unnecessary to send these indications, and allowing such unnecessary control frames to traverse the local channels would cause the local channels to optimize away less quickly. As it turns out, through some special magic dialplan code, it is possible to have a tones being played on a non-local channel, and it is important for the local channel to convey that the tones should be stopped. The result of having tones continue to be played on the non-local channel is that the tones play even once the channel has been bridged. By not blocking the -1 control frame type, we can ensure that this situation does not happen. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815	11 years ago
Joshua Colp	f1493f900e	audiohook: Read the correct number of samples based on audiohook format. ... Due to changes in audiohooks to support different sample rates the underlying storage of samples is in the format of the audiohook itself and not of the format being requested. This means that if a channel is using G722 the samples stored will be at 16kHz. If something subsequently reads from the audiohook at a format which is not the same sample rate as the audiohook the number of samples needs to be adjusted. Given the following example: 1. Channel writing into audiohook at 16kHz (as it is using G722). 2. Chanspy reading from audiohook at 8kHz. The original code would read 160 samples from the audiohook for each 20ms of audio. This is incorrect. Since the audio in the audiohook is at 16kHz the actual number needing to be read is 320. Failure to read this much would cause the audiohook to reset itself constantly as the buffer became full. This change adjusts the requested number of samples by determining the duration of audio requested and then calculating how many samples that would be in the audiohook format. ASTERISK-25247 #close Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d	11 years ago
Mark Michelson	d9094ddd73	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b	11 years ago
Richard Mudgett	875aee4c09	pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable. ... ASTERISK-25256 #close Reported by: Richard Mudgett Change-Id: I0b6be720b66fa956f6a798cd22ef8934eb0c0ff3	11 years ago
Richard Mudgett	e31cb6b248	strings.h: Fix issues with escape string functions. ... Fixes for issues with the ASTERISK-24934 patch. * Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is an empty string. If it were an empty string the functions returned NULL as if there were a memory allocation failure. This failure caused the AMI VarSet event to not get posted if the new value was an empty string. * Fixed dest buffer overwrite potential in ast_escape() and ast_escape_c(). If the dest buffer size is smaller than the space needed by the escaped s parameter string then the dest buffer would be written beyond the end by the nul string terminator. The num parameter was really the dest buffer size parameter so I renamed it to size. * Made nul terminate the dest buffer if the source string parameter s was an empty string in ast_escape() and ast_escape_c(). * Updated ast_escape() and ast_escape_c() doxygen function description comments to reflect reality. * Added some more unit test cases to /main/strings/escape to cover the empty source string issues. ASTERISK-25255 #close Reported by: Richard Mudgett Change-Id: Id77fc704600ebcce81615c1200296f74de254104	11 years ago
Matt Jordan	2e4bdbd78a	main/sorcery: Provide log messages when a wizard does not support an operation ... If a sorcery wizard does not support one of the 'optional' CRUD operations (namely the CUD), log a WARNING message so we are aware of why the operation failed. This also removes an assert in this case, as the CUD operation may have been triggered by an external system, in which case it is not a programming error but a configuration error. Change-Id: Ifecd9df946d9deaa86235257b49c6e5e24423b53	11 years ago
Mark Michelson	585d98fbb6	Merge "ARI: Added new functionality to get information on a single module." into 13	11 years ago
Mark Michelson	ca65ddcd19	Merge "bridge.c: Fixed race condition during attended transfer" into 13	11 years ago
Benjamin Ford	73e35d20de	ARI: Added new functionality to get information on a single module. ... An http request can be sent to retrieve information on a single module, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari /asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Information on a single module can now be retrieved ASTERISK-25173 Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463	11 years ago
Kevin Harwell	97ee0ee6c6	bridge.c: Fixed race condition during attended transfer ... During an attended transfer a thread is started that handles imparting the bridge channel. From the start of the thread to when the bridge channel is ready exists a gap that can potentially cause problems (for instance, the channel being swapped is hung up before the replacement channel enters the bridge thus stopping the transfer). This patch adds a condition that waits for the impart thread to get to a point of acceptable readiness before allowing the initiating thread to continue. ASTERISK-24782 Reported by: John Bigelow Change-Id: I08fe33a2560da924e676df55b181e46fca604577	11 years ago
Matt Jordan	6176ef782c	Merge "main/format_cap: Parse capabilities generated by ast_format_cap_get_names" into 13	11 years ago
Matt Jordan	bb76b88baf	main/sorcery: Don't fail object set creation from JSON if field fails ... Some individual fields may fail their conversion due to their default values being invalid for their custom handlers. In particular, configuration values that depend on others being enabled (and thus have an empty default value) are notorious for tripping this routine up. An example of this are any of the DTLS options for endpoints. Any of the DTLS options will fail to be applied (as DTLS is not enabled), causing the entire object set to be aborted. This patch makes it so that we log a debug message when skipping a field, and rumble on anyway. ASTERISK-25238 Change-Id: I0bea13de79f66bf9f9ae6ece0e94a2dc1c026a76	11 years ago
Matt Jordan	5f13c2226a	main/format_cap: Parse capabilities generated by ast_format_cap_get_names ... We have a strange relationship between the parsing of format capabilities from a string and their representation as a string. We expect the format capabilities to be expressed as a string in the following format: allow = !all,ulaw,alaw disallow = g722 While we would generate the string representation of those formats as: allow = (ulaw|alaw) disallow = (ulaw|alaw|g729...) When the configuration framework needs to store values as a string, it generates the format capabilities using the second representation; this representation however cannot be parsed when the entry is rehydrated. This patch fixes that by updating ast_format_cap_update_by_allow_disallow to parse an entry as if it were in the generated format if it has a leading '(' and a trailing ')'. ASTERISK-25238 Change-Id: I904d43caf4cf45af06f6aee0c9e58556eb91d6ca	11 years ago
Matt Jordan	328f0be806	main/devicestate: Prevent duplicate registration of device state providers ... Currently, the device state provider API will allow you to register a device state provider with the same case insensitive name more than once. This could cause strange issues, as the duplicate device state providers will not be queried when a device's state has to be polled. This patch updates the API such that a device state provider with the same name as one that has already registered will be rejected. Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2	11 years ago
Matt Jordan	e51d86682a	Merge "ARI: Added new functionality to get all module information." into 13	11 years ago
Benjamin Ford	47ea312b24	ARI: Added new functionality to get all module information. ... An http request can be sent to retrieve a list of all existing modules, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/ asterisk/modules" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Information on modules can now be retrieved Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0	11 years ago
Joshua Colp	b74b071369	res_sorcery_memory_cache: Backport to 13 ... Gerrit is complaining of conflicts when trying to create a patch series of all of the cherry-picked master commits, so I have instead squashed it all into one commit. ASTERISK-25067 #close Reported by: Matt Jordan Change-Id: I6dda90343fae24a75dc5beec84980024e8d61eb9	11 years ago
Richard Mudgett	ada7346792	res_pjsip: Need to use the same serializer for a pjproject SIP transaction. ... All send/receive processing for a SIP transaction needs to be done under the same threadpool serializer to prevent reentrancy problems inside pjproject and res_pjsip. * Add threadpool API call to get the current serializer associated with the worker thread. * Pick a serializer from a pool of default serializers if the caller of res_pjsip.c:ast_sip_push_task() does not provide one. This is a simple way to ensure that all outgoing SIP request messages are processed under a serializer. Otherwise, any place where a pushed task is done that would result in an outgoing out-of-dialog request would need to be modified to supply a serializer. Serializers from the default serializer pool are picked in a round robin sequence for simplicity. A side effect is that the default serializer pool will limit the growth of the thread pool from random tasks. This is not necessarily a bad thing. * Made pjsip_distributor.c save the thread's serializer name on the outgoing request tdata struct so the response can be processed under the same serializer. This is a cherry-pick from master. **** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a NOTE: session_inv_on_state_changed() is disassociating the dialog from the session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED. Unfortunately this is a tad too soon because our BYE request transaction has not completed yet. ASTERISK-25183 #close Reported by: Matt Jordan Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a	11 years ago
Matt Jordan	de3678c92e	Merge "sorcery/realtime: Add a bit of debug and warning messages for bad configs" into 13	11 years ago
Joshua Colp	43926568c3	Merge "rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format." into 13	11 years ago
Joshua Colp	c2f4bf142f	Merge "astfd: Fix buffer overflow in DEBUG_FD_LEAKS." into 13	11 years ago
Walter Doekes	b835312b4c	rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format. ... When running valgrind on Asterisk, it complained about: ==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304) ==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...) ==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292) ==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437) The code in question is a struct assignment, which may be performed by memcpy as a compiler optimization. It is changed to only copy the struct contents if source and destination are different. ASTERISK-25219 #close Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a	11 years ago
Walter Doekes	6551e16e03	astfd: Fix buffer overflow in DEBUG_FD_LEAKS. ... If DEBUG_FD_LEAKS was used and more file descriptors than the default of 1024 were available, some DEBUG_FD_LEAKS-patched functions would overwrite memory past the fixed-size (1024) fdleaks buffer. This change: - adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe - consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024 - stores pointers to constants instead of copying the contents - reorders the fdleaks struct for possibly tighter packing - adds a tiny bit of documentation ASTERISK-25212 #close Change-Id: Iacb69e7701c0f0a113786bd946cea5b6335a85e5	11 years ago
Scott Griepentrog	6b16fbfc22	Channel alert pipe: improve diagnostic error return ... When a frame is queued on a channel, any failure in ast_channel_alert_write is logged along with errno. This change improves the diagnostic message through aligning the errno value with actual failure cases. ASTERISK-25224 Reported by: Andrey Biglari Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b	11 years ago
Matt Jordan	8e07ab145d	sorcery/realtime: Add a bit of debug and warning messages for bad configs ... When a mapping does not exist between a sorcery.conf defined object and a realtime mapping in extconf, currently, the user will receive a slew of ERROR messages that don't really tell what is happening. Some ERROR messages may even be misleading, as they occur after the sorcery API has already given up on the attempt to load and create the sorcery object. This patch adds a bit of debug and a useful WARNING message for when a wizard's open callback fails for a particular object type. In the bad configurations that resulted in this patch, this provided a 'root cause' WARNING message that pointed in the right direction of the configuration problem. Change-Id: I1cc7344f2b015b8b9c85a7e6ebc8cb4753a8f80b	11 years ago
Matt Jordan	4c0b26b9e8	Merge "channel: Remove ignore of answer on non-outgoing channels." into 13	11 years ago
Matt Jordan	399cd8bcd9	main/pbx: Resolve case sensitivity regression in PBX hints ... When 8297136f was merged for ASTERISK-25040, a regression was introduced surrounding the case sensitivity of device names within hints. Previously, device names - such as 'sip/foo' - were compared in a case insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After that patch, only the case sensitive name would match, i.e., 'SIP/foo'. As a result, some dialplan hints stopped working. This patch re-introduces case insensitive matching for device names in hints. ASTERISK-25040 ASTERISK-25202 #close Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c (cherry picked from commit 96bbcf495a)	11 years ago
Matt Jordan	0a1700d286	Merge "threadpool, res_pjsip: Add serializer group shutdown API calls." into 13	11 years ago
Matt Jordan	e5499c3233	Merge "sorcery: Add ast_sorcery_object_unregister() API call." into 13	11 years ago
Richard Mudgett	84c12f9e0c	threadpool, res_pjsip: Add serializer group shutdown API calls. ... A module trying to unload needs to wait for all serializers it creates and uses to complete processing before unloading. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059	11 years ago
Joshua Colp	e4a2ef9e4e	channel: Remove ignore of answer on non-outgoing channels. ... Due to the way that channels can now be moved around inside of Asterisk it is possible for the outgoing flag of a channel to get cleared before it has been answered. This results in the bridge not receiving notification that the outgoing leg has been answered. This most easily exhibits itself with DTMF based blond transfers. Since the answer of the outgoing leg is ignored the other party continues to receive both a locally generated ringing and the media stream of the outgoing leg upon its answer. This results in no media being heard. This change removes the ignore of the answer and allows it to pass through. ASTERISK-25171 #close Change-Id: I82aedcec4f89f34a2e5472086dfc9a6c775bca8e	11 years ago
Richard Mudgett	20f3d77ab9	sorcery: Add ast_sorcery_object_unregister() API call. ... Find and unlink the specified sorcery object type to complement ast_sorcery_object_register(). Without this function you cannot completely unload individual modules that use sorcery for configuration. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88	11 years ago
Richard Mudgett	890c923786	AMI: Add Linkedid to the standard channel snapshot AMI event headers. ... * The AMI version is bumped to 2.8.0. ASTERISK-25189 #close Reported by: John Hardin Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac	11 years ago
Richard Mudgett	2602a7484b	test.c: Add unit test registration checks for summary and description. ... Added checks when a unit test is registered to see that the summary and description strings do not end with a new-line '\n' for consistency. The check generates a warning message and will cause the /main/test/registrations unit test to fail. * Updated struct ast_test_info member doxygen comments. Change-Id: I295909b6bc013ed9b6882e85c05287082497534d	11 years ago
Kevin Harwell	35a99b6394	bridge.c: Hangup attended transfer target if bridged ... After completing an attended transfer the transfer target channel was not being hung up after leaving the bridge. Added an explicit softhangup to hangup said channel, but only if it was previously bridged. ASTERISK-24782 #close Reported by: John Bigelow Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada	11 years ago
mjordan	de8c7f46ed	main/cdr: Carry over the disable flag when 'disable all' is specified ... The CDR_PROP function (as well as the NoCDR application) set the 'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This flag is supposed to be applied to all CDRs that are currently in the chain, as well as all CDRs that may be created in the future. Currently, however, the flag is only applied to the existing CDRs in the chain; new CDRs do not receive the 'disable all' flag. In particular, this affects parallel dials, which generate new CDRs for each pair of channels in the dial attempt. This patch carries over the 'disable all' flag when it is specified on a CDR and a new CDR is generated for the chain. ASTERISK-24344 #close Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e	11 years ago
Matt Jordan	78ea356e78	main/cdr: Copy context/exten on chained CDRs for parallel dials in subroutines ... When a parallel dial occurs, a new CDR will be created for each dial attempt that is made. In most circumstances, the act of creating each CDR in the chain will include a step that updates the Party A snapshot, which causes the context/extension of the Party A to be copied onto the CDR object. However, when the Party A is in a subroutine, we explicitly do *not* copy the context/extension onto the CDR. This prevents the Macro or GoSub routine name from blowing away the context/extension that the channel was originally executing in. For the original CDR, this is not a problem: the original CDR already recorded the last known 'good' state of the channel just prior to it going into the subroutine. However, for newly generated CDRs in a chain, there is no context/extension set on them. Since we are in a subroutine, we will never set the Party A's context/extension on the CDR, and we end up with a CDR with no destination recorded on it. This patch updates the creation of a chained CDR such that it copies over the original CDR's context/extension. This is the last known "good" state of the CDR, and is a reasonable starting point for the newly generated CDR. In the case where we are not in a subroutine, subsequent code will update the location of the CDR from the Party A information; in the case where we are in a subroutine, the context/extension on the original CDR is the correct information. ASTERISK-24443 #close Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a	11 years ago