26704 Commits (20ee33e22e3724c35ab3078c5032b87af5b4920c)

 

All Branches   

Search

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.	10 years ago
Mark Michelson	e28fbebc57	Merge "ARI: Rotate log channels." into 13	10 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	10 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	10 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	10 years ago
Richard Mudgett	e20f435b60	rtp_engine.h: Misc comment fixes. ... Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43	10 years ago
Richard Mudgett	bc5d7f9c37	chan_sip.c: Tweak glue->update_peer() parameter nil value. ... Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd	10 years ago
Richard Mudgett	13eb491e35	res_pjsip_session.c: Fix crashes seen when call cancelled. ... Two testsuite tests crashed in the same place as a result of an INVITE being CANCELed. tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp The session pointer is no longer in the inv->mod_data[session_module.id] location because the INVITE transaction has reached the terminated state. ASTERISK-25297 #close Reported by: Richard Mudgett Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427	10 years ago
Joshua Colp	9be856e3c6	Merge "Add a test event for inband ringing." into 13	10 years ago
Mark Michelson	48698a5e21	res_http_websocket: Properly encode 64 bit payload ... A test agent was continuously failing all ARI tests when run against Asterisk 13. As it turns out, the reason for this is that on those test runs, for some reason we decided to use the super extended 64 bit payload length for websocket text frames instead of the extended 16 bit payload length. For 64-bit payloads, the expected byte order over the network is 7, 6, 5, 4, 3, 2, 1, 0 However, we were sending the payload as 3, 2, 1, 0, 7, 6, 5, 4 This meant that we were saying to expect an absolutely MASSIVE payload to arrive. Since we did not follow through on this expected payload size, the client would sit patiently waiting for the rest of the payload to arrive until the test would time out. With this change, we use the htobe64() function instead of htonl() so that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a	10 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	10 years ago
Mark Michelson	c9099d06cc	Merge "holding_bridge: ensure moh participants get frames" into 13	10 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	10 years ago
Matt Jordan	f78a4b52b8	Bump the ARI version to 1.8.0 ... Due to backwards compatible changes, the ARI version should be bumped to 1.8.0 prior to the release of 13.5.0. Note that a previous patch already bumped the version of AMI for this release. Change-Id: I419033bfbbc0d3533a29ccb32b2981f39e0883e7	10 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	10 years ago
Joshua Colp	1997b6f677	Merge "res_pjsip: Add rtp_keepalive to sample config file." into 13	10 years ago
Mark Michelson	b4e19e414a	res_pjsip: Add rtp_keepalive to sample config file. ... Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19	10 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	10 years ago
Matt Jordan	4d8f47f4bf	Merge "audiohook: Use manipulated frame instead of dropping it." into 13	10 years ago
Joshua Colp	ff83c115c7	Merge "Local channels: Do not block control -1 payloads." into 13	10 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	10 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	10 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	10 years ago
Joshua Colp	d16347f33b	Merge "Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c" into 13	10 years ago
Rusty Newton	62c64c3bd1	Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c ... * In sip.conf.sample fix sentence where we said that WS or WSS are supported transports for use in an outbound register definition. They are not supported in that case. * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used to enable CDR on a channel. ASTERISK-24867 #close Reported by: Rusty Newton ASTERISK-24853 #close Reported by: PSDK Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca	10 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	10 years ago
Matt Jordan	34207887e6	Merge "chan_pjsip: Don't change formats when frame of unsupported format is received." into 13	10 years ago
Matt Jordan	8a1d9a9f83	Merge "res/res_musiconhold: Add a warning when MOH does not exist" into 13	10 years ago
Joshua Colp	29de5b497f	Merge "pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable." into 13	10 years ago
Michael Cargile	a23adcca3d	res/res_musiconhold: Add a warning when MOH does not exist ... Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b	10 years ago
Matt Jordan	03064daeb2	res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf ... Misconfiguring sorcery.conf with a 'config' wizard with no extra data will currently crash Asterisk on startup, as the wizard requires a comma delineated list to parse. This patch updates res_sorcery_config to check for the presence of the data before it starts manipulating it. Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847	10 years ago
Mark Michelson	1915aa5a54	Merge "sig_pri.h: force_restart_unavailable_chans in wrong scope" into 13	10 years ago
Joshua Colp	2c626ceb64	chan_pjsip: Don't change formats when frame of unsupported format is received. ... Receipt of an RTP packet currently causes the formats on an PJSIP channel to change to the format of the RTP packet. In some off-nominal cases it's possible for this to be a format that has not been configured or negotiated. This change makes it so only formats explicitly configured on the endpoint are allowed. ASTERISK-25258 #close Change-Id: If93d641fb6418a285928839300d7854cab8c1020	10 years ago
Patric Marschall	abb14ac5b8	sig_pri.h: force_restart_unavailable_chans in wrong scope ... In channels/sig_pri.h, struct sig_pri_span, the field force_restart_unavailable_chans is only defined if #if defined(HAVE_PRI_MCID) is true. All other occurences of force_restart_unavailable_chans are outside of the #if defined(HAVE_PRI_MCID) endif scope. ASTERISK-25257 #close Reported by: Patric Marschall Change-Id: I071de89cc2cd0d85927a013036e235851f672549	10 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	10 years ago
Matt Jordan	9d7f689b4b	Merge "ARI: Add support for push configuration of dynamic object" into 13	10 years ago
Matt Jordan	325d83f37f	Merge "strings.h: Fix issues with escape string functions." into 13	10 years ago
Matt Jordan	8bcf6d2801	ARI: Add support for push configuration of dynamic object ... This patch adds support for push configuration of dynamic, i.e., sorcery, objects in Asterisk. It adds three new REST API calls to the 'asterisk' resource: * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current object given its ID. This returns back a list of ConfigTuples, which define the fields and their present values that make up the object. * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an object. A body may be passed with the request that contains fields to populate in the object. The same format as what is retrieved using the GET operation is used for the body, save that we specify that the list of fields to update are contained in the "fields" attribute. * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic object from its backing storage. Note that the success/failure of these operations is somewhat configuration dependent, i.e., you must be using a sorcery wizard that supports the operation in question. If a sorcery wizard does not support the create or delete mechanisms, then the REST API call will fail with a 403 forbidden. ASTERISK-25238 #close Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c	10 years ago
Matt Jordan	80eaf0b025	Merge "res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer()." into 13	10 years ago
Matt Jordan	174f0e9d4d	Merge "res_pjsip_session.c: Add some helpful comments and minor tweaks." into 13	10 years ago
Matt Jordan	bba2c44ac4	Merge "res_pjsip_session.c: Fix off nominal crash potential in debug message." into 13	10 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	10 years ago
Richard Mudgett	243c0d1609	parking_applications.c: Fix ast_verb() line terminator. ... Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775	10 years ago
Richard Mudgett	c782320c68	res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park. ... setup_park_common_datastore() was assuming that a non-NULL string returned for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty strings. Things got crashy as a result. * Made setup_park_common_datastore() treat the channel variable values the same whether they are NULL or empty for ATTENDEDTRANSFER and BLINDTRANSFER. ASTERISK-25254 #close Reported by: Richard Mudgett Change-Id: I9a9c174b33f354f35f82cc6b7cea8303adbaf9c2	10 years ago
Richard Mudgett	2735dd5b2d	res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer(). ... Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb	10 years ago
Richard Mudgett	3d0ca343ca	res_pjsip_session.c: Add some helpful comments and minor tweaks. ... Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743	10 years ago
Richard Mudgett	8d08bb179c	res_pjsip_session.c: Fix off nominal crash potential in debug message. ... Change-Id: I09928297927ee85f7655289acee3a586816466bc	10 years ago
Joshua Colp	5086bdacfb	Merge "apps/app_dictate: Fix typo in attribution" into 13	10 years ago
Mark Michelson	ca41785774	Merge "ARI: Fixed unload mode for unload module." into 13	10 years ago
Matt Jordan	0a1a550593	apps/app_dictate: Fix typo in attribution ... Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian (GameGamer43) for pointing that out. Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106	10 years ago