4859 Commits (16.5)

Author	SHA1	Message	Date
Kevin Harwell	3e794ada8e	AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media ... After receiving a 200 OK with a declined stream in response to a T.38 initiated re-invite Asterisk would crash when attempting to dereference a NULL session media object. This patch checks to make sure the session media object is not NULL before attempting to use it. ASTERISK-28495 patches: ast-2019-004.patch submitted by Alexei Gradinari (license 5691) Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572	7 years ago
Joshua Colp	1756029237	res_rtp_asterisk: Move where DTLS MTU variable is defined. ... The DTLS MTU variable is not dependent on pjproject and should not exist in its block. Change-Id: I7e97d64dc192f2ac81bfe2b72b8229d321c7d026	7 years ago
George Joseph	2126dc3021	res_pjsip_messaging: Check for body in in-dialog message ... We now check that a body exists and it has a length > 0 before attempting to process it. ASTERISK-28447 Reported-by: Gil Richard Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f	7 years ago
Kevin Harwell	83aba363fe	res_pjsip_sdp_rtp: Remove unused variable ... The variable 'endpoint_caps' in function 'set_caps' is not used, so remove. ASTERISK-28458 Change-Id: Ia8766d05a0738aecb29dd018302c2dafca5cab34	7 years ago
Friendly Automation	635affeac5	Merge "res_fax: gateway sends T.38 request to both endpoints if V.21 detected" into 16	7 years ago
Joshua Colp	82789aafd6	res_rtp_asterisk: Add support for DTLS packet fragmentation. ... This change adds support for larger TLS certificates by allowing OpenSSL to fragment the DTLS packets according to the configured MTU. By default this is set to 1200. This is accomplished by implementing our own BIO method that supports MTU querying. The configured MTU is returned to OpenSSL which fragments the packet accordingly. When a packet is to be sent it is done directly out the RTP instance. ASTERISK-28018 Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06	7 years ago
Alexei Gradinari	6321b559b9	res_fax: gateway sends T.38 request to both endpoints if V.21 detected ... According T.38 Gateway 'Use case 3' https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway T.38 Gateway should send T.38 negotiation request to called endpoint if FAX preamble (using V.21 detector) generated by called endpoint. But it does not, because fax_gateway_detect_v21 constructs T.38 negotiation request, but forwards it only to other channel, not to the channel on which FAX preamble is detected. Some SIP endpoints could be improperly configured to rely on the other side to initiate T.38 re-INVITEs. With this patch the T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38. Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39	7 years ago
Joshua Colp	de38c9c3b3	Merge "res_fax: fix segfault on inactive "reserved" fax session" into 16	7 years ago
Alexei Gradinari	e77704f45c	res_fax: add channel name to CLI 'fax show session' ... This patch adds a channel name to output of CLI 'fax show session' and also expands the channel name field up to 30 characters on CLI 'fax show sessions' Change-Id: Id059c43ff41811f5e76712b83fb63b8f246da953	7 years ago
Alexei Gradinari	e0a574253e	res_fax: fix segfault on inactive "reserved" fax session ... The change #10017 "Handle fax gateway being started more than once" introdiced a bug which leads to segfault in res_fax_spandsp. The res_fax_spandsp module does not support reserving sessions, so fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE. The fax_gateway_start does not create a real fax session if the fax session is already present and the state is not AST_FAX_STATE_RESERVED. But the "reserved" session created for res_fax_spandsp has state AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting. Then when fax_gateway_framehook is called and gateway T.38 state is NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to segfault, because session tech_pvt is not set, i.e. the tech session was not initialized/started. This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved" session created for res_fax_spandsp will start. This patch also adds extra check and log ERROR if tech_pvt is not set before call tech->write. ASTERISK-27981 #close Change-Id: Ife3e65e5f18c902db2ff0538fccf7d28f88fa803	7 years ago
Friendly Automation	0fc6617246	Merge "res_rtp_asterisk: timestamp should be unsigned instead of signed int" into 16	7 years ago
Morten Tryfoss	9351aa3f0e	res_rtp_asterisk: timestamp should be unsigned instead of signed int ... Using timestamp with signed int will cause timestamps exceeding max value to be negative. This causes the jitterbuffer to do passthrough of the packet. ASTERISK-28421 Change-Id: I9dabd0718180f2978856c50f43aac4e52dc3cde9	7 years ago
George Joseph	79b15d0b30	res_rtp_asterisk: Add ability to propose local address in ICE ... You can now add the "include_local_address" flag to an entry in rtp.conf "[ice_host_candidates]" to include both the advertized address and the local address in ICE negotiation: [ice_host_candidates] 192.168.1.1 = 1.2.3.4,include_local_address This causes both 192.168.1.1 and 1.2.3.4 to be advertized. Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db	7 years ago
Joshua Colp	2aa9bc6d2c	Merge "res_rtp_asterisk: Fix sequence number cycling and packet loss count." into 16	7 years ago
Joshua Colp	cf2f8db1b7	Merge "pjsip_options.c: Allow immediate qualifies for new contacts." into 16	7 years ago
Joshua Colp	ece29db9bd	res_rtp_asterisk: Fix sequence number cycling and packet loss count. ... This change fixes two bugs which both resulted in the packet loss count exceeding 65,000. The first issue is that the sequence number check to determine if cycling had occurred was using the wrong variable resulting in the check never seeing that cycling has occurred, throwing off the packet loss calculation. It now uses the correct variable. The second issue is that the packet loss calculation assumed that the received number of packets in an interval could never exceed the expected number. In practice this isn't true due to delayed or retransmitted packets. The expected will now be updated to the received number if the received exceeds it. ASTERISK-28379 Change-Id: If888ebc194ab69ac3194113a808c414b014ce0f6	7 years ago
Ben Ford	941dead08d	pjsip_options.c: Allow immediate qualifies for new contacts. ... When multiple endpoints try to register close together using the same AOR with qualify_frequency set, one contact would qualify immediately while the other contacts would have to wait out the duration of the timer before being able to qualify. Changing the conditional to check the contact container count for a non-zero value allows all contacts to qualify immediately. Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415	7 years ago
agupta	9a0fa51443	stasis: Hangup channel for Local channel No such extension error ... When we use early bridge with create and dial from stasis using Local channel and the dialplan does not any entry the it is returned from core_local.c with No such extension . In such case asterisk locks up till the channel is not hangup with the error Exceptionally long voice queue length * Found that in such case app_control_dial fails on ast_call method and return -1 * Since it is called from stasis_app_send_command_async and return -1 does not cause resources to be freed and since no PBX exist it is not able to read from channel causing exceptionally long queue * After putting this code found that the channel was releasing immediately and resources were freed. ASTERISK-28399 Reported by: Abhay Gupta Tested by: Abhay Gupta Change-Id: I0a55c923fc6995559f808d63b9488762b4489318	7 years ago
Friendly Automation	27696cbda6	Merge "stasis: Only place stasis created and dialed channels into dial bridge." into 16	7 years ago
Friendly Automation	7bddfdbfa6	Merge "rtp: Add support for transport-cc in receiver direction." into 16	7 years ago
George Joseph	5002169d6a	res_pjsip: Check return from pjsip_parse_uri calls ... Updated ast_sip_create_rdata_with_contact and registrar_find_contact to check the return from pjsip_parse_uri before attempting to use the uri returned. ASTERISK-28402 Reported-by: Ross Beer Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7	7 years ago
agupta	39c5188bec	stasis: Only place stasis created and dialed channels into dial bridge. ... The dial bridge is meant to hold channels which have been created and dialed in stasis. It handles the frames coming from them and raises the appropriate events. It was possible for the code to mistakenly place calls which came from the dialplan into the dial bridge if they were not in an answered state. These channels are not outgoing channels and should not be placed into the dial bridge. The code now checks to ensure that only stasis created channels are placed into the dial bridge by checking that a PBX does not exist on the channel. ASTERISK-27756 Change-Id: Ideee69ff06c9a0b31f7ed61165f5c055f51d21b6	7 years ago
Joshua Colp	5023f02b2d	rtp: Add support for transport-cc in receiver direction. ... The transport-cc draft is a mechanism by which additional information about packet reception can be provided to the sender of packets so they can do sender side bandwidth estimation. This is accomplished by having a transport specific sequence number and an RTCP feedback message. This change implements this in the receiver direction. For each received RTP packet where transport-cc is negotiated we store the time at which the RTP packet was received and its sequence number. At a 1 second interval we go through all packets in that period of time and use the stored time of each in comparison to its preceding packet to calculate its delta. This delta information is placed in the RTCP feedback message, along with indicators for any packets which were not received. The browser then uses this information to better estimate available bandwidth and adjust accordingly. This may result in it lowering the available send bandwidth or adjusting how "bursty" it can be. ASTERISK-28400 Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc	7 years ago
Kevin Harwell	e3a758975d	mwi core: Move core MWI functionality into its own files ... There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0	7 years ago
Friendly Automation	74d79bdf71	Merge "res_pjsip: Added a norefersub configuration setting" into 16	7 years ago
Friendly Automation	93d36953fb	Merge "res_remb_modifier: Propertly initialize bitrate to 0.0" into 16	7 years ago
George Joseph	7487fc88d2	res_remb_modifier: Propertly initialize bitrate to 0.0 ... ...and return the frame unaltered if bitrate can't be determined. Change-Id: Ib2175ab84f85a3d7060d31625f5a2c7fbcc2ba4c	7 years ago
Dan Cropp	eca8c440d2	res_pjsip: Added a norefersub configuration setting ... Added a new PJSIP global setting called norefersub. Default is true to keep support working as before. res_pjsip_refer: Configures PJSIP norefersub capability accordingly. Checks the PJSIP global setting value. If it is true (default) it adds the norefersub capability to PJSIP. If it is false (disabled) it does not add the norefersub capability to PJSIP. This is useful for Cisco switches that do not follow RFC4488. ASTERISK-28375 #close Reported-by: Dan Cropp Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9	7 years ago
Sean Bright	022e784b7a	res_mwi_devstate: Specify AST_MODFLAG_LOAD_ORDER to enable load priority ... Suggested by abelbeck on the issue tracker. ASTERISK~28384 Reported by: abelbeck Change-Id: Icee0fff2b58dbfaa80f2b68270fe69dfb0463fc0	7 years ago
Joshua Colp	7a6895baf5	Merge "res_ael: Use Gosub in for loop expressions" into 16	7 years ago
Joshua Colp	d734e1bbfa	Merge "ARI: Run 'make ari-stubs'" into 16	7 years ago
Joshua Colp	95dac45148	Merge "res_ael: Fix pattern matching against literal '+'" into 16	7 years ago
George Joseph	898765d919	ARI: Run 'make ari-stubs' ... An earlier contributor apparently forgot to run 'make ari-stubs' before committing after making ARI model changes. Change-Id: I7813e5638e2821d11f4b968dc2aeab4f725190a6	7 years ago
Sean Bright	116dc9c9b3	res_ael: Create consistent label names across reloads ... Reset the internal counter that the AEL2 compiler uses for unique label names before compiling. This keeps dialplan labels consistent across reloads assuming the AEL2 has not changed. ASTERISK-17799 #close Reported by: Kirill Katsnelson Change-Id: I30b3cc887d1ee0644d3f341e2fef16f525d7fae5	7 years ago
Sean Bright	ea3109beaa	res_ael: Use Gosub in for loop expressions ... In AEL2, if a 'for' statement contains macro* calls, like: for (&iterator(${TRY},A); "${A}" != ""; &iterate(A)) { The AEL2 parser will translate these into calls to the deprecated Macro dialplan application and use the antiquated pipe delimiter. Instead, convert these into calls to the Gosub dialplan application and use commas as argument separators. ASTERISK-18593 #close Reported by: Luke-Jr * 'macro' in this context means AEL2 macros, not the 'Macro' application Change-Id: I3d73716033b8e3e42e0209d355bf5f10c97045fc	7 years ago
Sean Bright	71c7864d1d	res_ael: Fix pattern matching against literal '+' ... When generating the regular expression that matches against existing extensions, we need to escape literal characters that can also be regular expression metacharacters. This was already being done for '*' but we need to do the same for '+'. In passing, remove some unreachable code - strcmp() is already run immediately when entering extension_matches(). ASTERISK-14939 #close Reported by: klaus3000 Change-Id: I8d2cccb3479168fba1b0a6704c52198b396468f1	7 years ago
Alexei Gradinari	acfbfef8ad	res_pjsip: Fix transport_states ref leak ... Add missing ao2_ref(transport_state, -1) while iterate on a transport_states container. Change-Id: I40e35b5a339121300c80075c30db47201a6c374e	7 years ago
Friendly Automation	777ee8290a	Merge "main/json.c: Added app_name, app_data to channel type" into 16	7 years ago
Friendly Automation	90781674ad	Merge "res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics" into 16	7 years ago
sungtae kim	d5a318f148	main/json.c: Added app_name, app_data to channel type ... It was difficult to check the channel's current application and parameters using ARI for current channels. Added app_name, app_data items to show the current application information. ASTERISK-28343 Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c	7 years ago
Joshua Colp	4f0b8c3ed3	Merge "bridge_softmix: use a float type to store the internal REMB bitrate" into 16	7 years ago
Kevin Harwell	94eeba6147	bridge_softmix: use a float type to store the internal REMB bitrate ... REMB's exponent is 6-bits (0..63) and has a mantissa of 18-bits. We were using an unsigned integer to represent the bitrate. However, that type is not large enough to hold all potential bitrate values. If the bitrate is large enough bits were being shifted off the "front" of the mantissa, which caused the wrong value to be sent to the browser. This patch makes it so it now uses a float type to hold the bitrate. Using a float allows for all bitrate values to be correctly represented. ASTERISK-28255 Change-Id: Ice00fdd16693b16b41230664be5d9f0e465b239e	7 years ago
Matthew Fredrickson	ae1aeb930e	res/res_rtp_asterisk: Enable rxjitter calculation for video ... It looks like we're not properly calculating jitter values on received video streams. This patch enables the code that does jitter calculations for those streams. Change-Id: Iaac985808829c8f034db8c57318789c4c8c11392	7 years ago
sungtae kim	bbc13b1f1f	res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics ... Added ARI resource for channel statistics. GET /ari/channels/{channelId}/rtp_statistics : It returns given channel's rtp statistics detail. ASTERISK-28320 Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376	7 years ago
Friendly Automation	03c5b916ff	Merge "res/res_ari: Added timestamp as a requirement for all ARI events" into 16	7 years ago
sungtae kim	6d455487d9	res/res_ari: Added timestamp as a requirement for all ARI events ... Changed to requirement to having timestamp for all of ARI events. The below ARI events were changed to having timestamp. PlaybackStarted, PlaybackContinuing, PlaybackFinished, RecordingStarted, RecordingFinished, RecordingFailed, ApplicationReplaced, ApplicationMoveFailed ASTERISK-28326 Change-Id: I382c2fef58f5fe107e1074869a6d05310accb41f	8 years ago
Alexei Gradinari	634314c727	res_config_odbc: set empty extended field as a single whitespace ... If Realtime @ variable value is NULL or empty or contains only whitespaces then when we try to retrieve it using PJSIP_ENDPOINT we get WARNING pjsip_endpoint_function_read: Unknown property @my_var for PJSIP endpoint. And the variable is missing in the result of CLI pjsip show endpoint. This patch keeps empty sorcery extended field. ASTERISK-28341 #close Change-Id: I221fccc04cbfa2be17ce971f64ae0e74e465eea0	8 years ago
George Joseph	396dac8888	Merge "res/res_stasis: Fixed wrong StasisEnd timestamp" into 16	8 years ago
sungtae kim	9e36707dca	res/res_stasis: Fixed wrong StasisEnd timestamp ... Because StasisEnd's timestamp created it's own timestamp, it makes wrong timestamp, compare to other channel event(ChannelDestroyed). Fixed to getting a timestamp from the Channel's timestamp. ASTERISK-28333 Change-Id: I5eb8380fc472f1100535a6bc4983c64767026116	8 years ago
George Joseph	f5f977baec	app.c: Remove deletion of pool topic on mwi state delete ... As part of an earlier voicemail refactor, ast_delete_mwi_state_full was modified to remove the pool topic for a mailbox when the state was deleted. This was an attempt to prevent stale topics from accumulating when app_voicemail was reloaded and a mailbox went away. Unfortunately because of the fact that when app_voicemail reloads, ALL mailboxes are deleted then only current ones recreated, topics were being removed from the pool that still had subscribers on them, then recreated as new topics of the same name. So now modules like res_pjsip_mwi are listening on a topic that will never receive any messages because app_voicemail is publishing on a different topic that happens to have the same name. The solutiuon to this is not easy and given that accumulating topics for deleted mailboxes is less evil that not sending NOTIFYs... * Removed the call to stasis_topic_pool_delete_topic in ast_delete_mwi_state_full. Also: * Fixed a topic reference leak in res_pjsip_mwi mwi_stasis_subscription_alloc. * Added some debugging to mwi_stasis_subscription_alloc, stasis_topic_create, and topic_dtor. * Fixed a topic reference leak in an error path in internal_stasis_subscribe. ASTERISK-28306 Reported-by: Jared Hull Change-Id: Id7da0990b3ac4be4b58491536b35f41291247b27	8 years ago