4889 Commits (18f5f5fc99a8106c1856d13aeeeeafa2b13ef033)

Author	SHA1	Message	Date
Kevin Harwell	18f5f5fc99	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	6 years ago
Kevin Harwell	172e183b9d	res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions ... res_pjsip_mwi allows both solicited and unsolicited MWI subscription types. While both can be set in the configuration for a given endpoint/aor, only one is allowed. Precedence is given to unsolicited. Meaning if an endpoint/aor is configured to allow both types then the solicited subscription is rejected when it comes in. However, there is a configuration option to override that behavior: mwi_subscribe_replaces_unsolicited When set to "yes" then when a solicited subscription comes in instead of rejecting it Asterisk is suppose to replace the unsolicited one if it exists. Prior to this patch there was a bug in Asterisk that allowed the solicted one to be added, but did not remove the unsolicited. As a matter of fact a new unsolicited subscription got added everytime a SIP register was received. Over time this eventually could "flood" a phone with SIP notifies. This patch fixes that behavior to now make it work as expected. If configured to do so a solicited subscription now properly replaces the unsolicited one. As well when an unsubscribe is received the unsolicited subscription is restored. Logic was also put in to handle reloads, and any configuration changes that might result from that. For instance, if a solicited subscription had previously replaced an unsolicited one, but after reload it was configured to not allow that then the solicited one needs to be shutdown, and the unsolicited one added. ASTERISK-28488 Change-Id: Iec2ec12d9431097e97ed5f37119963aee41af7b1	6 years ago
Alexei Gradinari	3ef52b0b17	Fix misname 'res_external_mwi' to 'res_mwi_external' in comments. ... Change-Id: Ic784be8500e5cb75dcb34bae9f03cfd93b6b34fb	6 years ago
George Joseph	a145669534	Merge "res_pjsip: Channel variable SIPFROMDOMAIN"	6 years ago
George Joseph	8da4e28a81	res_ari.c: Prefer exact handler match over wildcard ... Given the following request path and 2 handler paths... Request: /channels/externalMedia Handler: /channels/{channelId} "wildcard" Handler: /channels/externalmedia "non-wildcard" ...if /channels/externalMedia was registered as a handler after /channels/{channelId} as shown above, the request would automatically match the wildcard handler and attempt to parse "externalMedia" into the channelId variable which isn't what was intended. It'd work if the non-wildard entry was defined in rest-api/api-docs/channels.json before the wildcard entry but that makes the json files order-dependent which isn't a good thing. To combat this issue, the search loop saves any wildcard match but continues looking for exact matches at the same level. If it finds one, it's used. If it hasn't found an exact match at the end of the current level, the wildcard is used. Regardless, after searching the current level, the wildcard is cleared so it won't accidentally match for a different object or a higher level. BTW, it's currently not possible for more than 1 wildcard entry to be defined for a level. For instance, there couldn't be: Handler: /channels/{channelId} Handler: /channels/{channelName} We wouldn't know which one to match. Change-Id: I574aa3cbe4249c92c30f74b9b40e750e9002f925	6 years ago
Stas Kobzar	c7270dca81	res_pjsip: Channel variable SIPFROMDOMAIN ... In chan_sip, there was variable SIPFROMDOMAIN that allows to set From header URI domain per channel. This patch introduces res_pjsip variable SIPFROMDOMAIN for backward compatibility with chan_sip. ASTERISK-28489 Change-Id: I715133e43172ce2a1e82093538dc39f9e99e5f2e	6 years ago
Kevin Harwell	b805e1237d	srtp: Fix possible race condition, and add NULL checks ... Somehow it's possible for the srtp session object to be NULL even though the Asterisk srtp object itself is valid. When this happened it would cause a crash down in the srtp code when attempting to protect or unprotect data. After looking at the code there is at least one spot that makes this situation possible. If Asterisk fails to unprotect the data, and after several retries it still can't then the srtp->session gets freed, and set to NULL while still leaving the Asterisk srtp object around. However, according to the original issue reporter this does not appear to be their situation since they found no errors logged stating the above happened (which Asterisk does for that situation). An issue was found however, where a possible race condition could occur between the pjsip incoming negotiation, and the receiving of RTP packets. Both places could attempt to create/setup srtp for the same rtp instance at the same time. This potentially could be the cause of the problem as well. Given the above this patch adds locking around srtp setup for a given rtp, or rtcp instance. NULL checks for the session have also been added within the protect and unprotect functions as a precaution. These checks should at least stop Asterisk from crashing if it gets in this situation again. This patch also fixes one other issue noticed during investigation. When doing a replace the old object was freed before creating the replacement. If the new replacement object failed to create then the rtp/rtcp instance would now point to freed srtp data which could potentially cause a crash as well when the next attempt to reference it was made. This is now fixed so the old srtp object is kept upon replacement failure. Lastly, more logging has been added to help diagnose future issues. ASTERISK-28472 Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc	6 years ago
Friendly Automation	e5ee1b04c9	Merge "res_musiconhold: Use a vector instead of custom array allocation"	6 years ago
Sean Bright	1f8ae708a0	res_musiconhold: Use a vector instead of custom array allocation ... Change-Id: Ic476a56608b1820ca93dcf68d10cd76fc0b94141	6 years ago
Joshua Colp	86452c9fa4	res_pjsip: Fix multiple of the same contact in "pjsip show contacts". ... The code for gathering contacts could result in the same contact being retrieved and added to the list multiple times. The container which stores the contacts to display will now only allow a contact to be added to it once instead of multiple times. ASTERISK-28228 Change-Id: I805185cfcec03340f57d2b9e6cc43c49401812df	6 years ago
Sean Bright	7ce9ee7f2e	res_musiconhold: Use ast_pipe_nonblock() wrapper ... Change-Id: Ib0a4b41e5ececbe633079e2d8c2b66c031d2d1f2	6 years ago
Sean Bright	2424ecaf66	res_config_sqlite3: Only join threads that we started ... ASTERISK-28477 #close Reported by: Dennis ASTERISK-28478 #close Reported by: Dennis Change-Id: I77347ad46a86dc5b35ed68270cee56acefb4f475	6 years ago
Joshua Colp	2feac1d361	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	6 years ago
Kevin Harwell	857ee76f4b	Merge "MWI: Update modules that subscribe to MWI to use new API calls"	6 years ago
George Joseph	3c520147e1	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	6 years ago
Kevin Harwell	9637e1dfdc	MWI: Update modules that subscribe to MWI to use new API calls ... The MWI core recently got some new API calls that make tracking MWI state lifetime more reliable. This patch updates those modules that subscribe to specific MWI topics to use the new API. Specifically, these modules now subscribe to both MWI topics and MWI state. ASTERISK-28442 Change-Id: I32bef880b647246823dbccdf44a98d384fcabfbd	6 years ago
Kevin Harwell	93936e367d	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	6 years ago
sungtae kim	613a335de5	res/ari/resource_channels.c: Added hangup reason code for channels ... Currently, DELETE /ari/channels/<channelID> supports only few hangup reasons. It's good enough for simple use, but when it needs to set the detail reason, it comes challenges. Added reason_code query parameter for that. ASTERISK-28385 Change-Id: I1cf1d991ffd759d0591b347445a55f416ddc3ff2	6 years ago
Alexei Gradinari	f414ca069c	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	6 years ago
Joshua Colp	a8e5cf557d	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	6 years ago
Friendly Automation	03783cfa1b	Merge "res_fax: fix segfault on inactive "reserved" fax session"	6 years ago
Alexei Gradinari	1b62781be0	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	6 years ago
Alexei Gradinari	bfd93995d9	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	6 years ago
Friendly Automation	9b0a21c402	Merge "res_rtp_asterisk: timestamp should be unsigned instead of signed int"	6 years ago
Friendly Automation	522303681c	Merge "res_rtp_asterisk: Add ability to propose local address in ICE"	6 years ago
Matt Jordan	0bb38796b7	res_prometheus: Add metrics for PJSIP outbound registrations ... When monitoring Asterisk instances, it's often useful to know when an outbound registration fails, as this often maps to the notion of a trunk and having a trunk fail is usually a "bad thing". As such, this patch adds monitoring metrics that track the state of PJSIP outbound registrations. It does this by looking for the Registry events coming across the Stasis system topic, and publishing those as metrics to Prometheus. Note that while this may support other outbound registration types (IAX2, SIP, etc.) those haven't been tested. Your mileage may vary. (And why are you still using IAX2 and SIP? It's 2019 folks. Get with the program.) This patch also adds Sorcery observers to handle modifications to the underlying PJSIP outbound registration objects. This is useful when a reload is triggered that modifies the properties of an outbound registration, or when ARI push configuration is used and an object is updated or deleted. Because we rely on properties of the registration object to define the metric (label key/value pairs), we delete the relevant metric when we notice that something has changed and wait for a new Stasis message to arrive to re-create the metric. ASTERISK-28403 Change-Id: If01420e38530fc20b6dd4aa15cd281d94cd2b87e	6 years ago
Matt Jordan	a2648b22eb	res_prometheus: Add CLI commands ... This patch adds a few CLI commands to the res_prometheus module to aid system administrators setting up and configuring the module. This includes: * prometheus show status: Display basic statistics about the Prometheus module, including its essential configuration, when it was last scraped, and how long the scrape took. The last two bits of information are useful when Prometheus isn't generating metrics appropriately, as it will at least tell you if Asterisk has had its HTTP route hit by the remote server. * prometheus show metrics: Dump the current metrics to the CLI. Useful for system administrators to see what metrics are currently available without having to cURL or go to Prometheus itself. ASTERISK-28403 Change-Id: Ic09813e5e14b901571c5c96ebeae2a02566c5172	6 years ago
Matt Jordan	066280f0cc	res_prometheus: Add Asterisk bridge metrics ... This patch adds basic Asterisk bridge statistics to the res_prometheus module. This includes: * asterisk_bridges_count: The current number of bridges active on the system. * asterisk_bridges_channels_count: The number of channels active in a bridge. In all cases, enough information is provided with each bridge metric to determine a unique instance of Asterisk that provided the data, along with the technology, subclass, and creator of the bridge. ASTERISK-28403 Change-Id: Ie27417dd72c5bc7624eb2a7a6a8829d7551788dc	6 years ago
Matt Jordan	ed6cd13b5b	res_prometheus: Add Asterisk endpoint metrics ... This patch adds basic Asterisk endpoint statistics to the res_prometheus module. This includes: * asterisk_endpoints_state: The current state (unknown, online, offline) for each defined endpoint. * asterisk_endpoints_channels_count: The current number of channels associated with a given endpoint. * asterisk_endpoints_count: The current number of defined endpoints. In all cases, enough information is provided with each endpoint metric to determine a unique instance of Asterisk that provided the data, as well as the underlying technology and resource definition. ASTERISK-28403 Change-Id: I46443963330c206a7d12722d08dcaabef672310e	6 years ago
Morten Tryfoss	3224ac07c9	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	6 years ago
Matt Jordan	0760af71ad	res_prometheus: Add Asterisk channel metrics ... This patch adds basic Asterisk channel statistics to the res_prometheus module. This includes: * asterisk_calls_sum: A running sum of the total number of processed calls * asterisk_calls_count: The current number of calls * asterisk_channels_count: The current number of channels * asterisk_channels_state: The state of any particular channel * asterisk_channels_duration_seconds: How long a channel has existed, in seconds In all cases, enough information is provided with each channel metric to determine a unique instance of Asterisk that provided the data, as well as the name, type, unique ID, and - if present - linked ID of each channel. ASTERISK-28403 Change-Id: I0db306ec94205d4f58d1e7fbabfe04b185869f59	6 years ago
Friendly Automation	12cad5ec1a	Merge "Add core Prometheus support to Asterisk"	6 years ago
Matt Jordan	c50f29dfad	Add core Prometheus support to Asterisk ... Prometheus is the defacto monitoring tool for containerized applications. This patch adds native support to Asterisk for serving up Prometheus compatible metrics, such that a Prometheus server can scrape an Asterisk instance in the same fashion as it does other HTTP services. The core module in this patch provides an API that future work can build on top of. The API manages metrics in one of two ways: (1) Registered metrics. In this particular case, the API assumes that the metric (either allocated on the stack or on the heap) will have its value updated by the module registering it at will, and not just when Prometheus scrapes Asterisk. When a scrape does occur, the metrics are locked so that the current value can be retrieved. (2) Scrape callbacks. In this case, the API allows consumers to be called via a callback function when a Prometheus initiated scrape occurs. The consumers of the API are responsible for populating the response to Prometheus themselves, typically using stack allocated metrics that are then formatted properly into strings via this module's convenience functions. These two mechanisms balance the different ways in which information is generated within Asterisk: some information is generated in a fashion that makes it appropriate to update the relevant metrics immediately; some information is better to defer until a Prometheus server asks for it. Note that some care has been taken in how metrics are defined to minimize the impact on performance. Prometheus's metric definition and its support for nesting metrics based on labels - which are effectively key/value pairs - can make storage and managing of metrics somewhat tricky. While a naive approach, where we allow for any number of labels and perform a lot of heap allocations to manage the information, would absolutely have worked, this patch instead opts to try to place as much information in length limited arrays, stack allocations, and vectors to minimize the performance impacts of scrapes. The author of this patch has worked on enough systems that were driven to their knees by poor monitoring implementations to be a bit cautious. Additionally, this patch only adds support for gauges and counters. Additional work to add summaries, histograms, and other Prometheus metric types may add value in the future. This would be of particular interest if someone wanted to track SIP response types. Finally, this patch includes unit tests for the core APIs. ASTERISK-28403 Change-Id: I891433a272c92fd11c705a2c36d65479a415ec42	6 years ago
George Joseph	be83591f99	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
Friendly Automation	0575a8923b	Merge "res_rtp_asterisk: Fix sequence number cycling and packet loss count."	7 years ago
Friendly Automation	c0640a3033	Merge "pjsip_options.c: Allow immediate qualifies for new contacts."	7 years ago
Joshua Colp	7a6fd83aca	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	86836e0442	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	85242a9bb9	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
Joshua Colp	8c1680fce4	Merge "stasis: Only place stasis created and dialed channels into dial bridge."	7 years ago
Friendly Automation	249be1d7ae	Merge "rtp: Add support for transport-cc in receiver direction."	7 years ago
George Joseph	ef92c69fa8	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	71040078a3	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	6bb70c93f1	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	ff0d0ac23a	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
Joshua Colp	9c070a4ae3	Merge "res_pjsip: Added a norefersub configuration setting"	7 years ago
George Joseph	d4e25710f7	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	cffa2a74cb	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	e69fcdfd83	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	adc8aa8125	Merge "res_ael: Use Gosub in for loop expressions"	7 years ago