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Author	SHA1	Message	Date
George Joseph	f3e5419d41	app_confbridge: Attended transfer event fixup ... When a channel already in a conference bridge is attended transfered to another extension, or when an existing call is attended transferred into a conference bridge, we now generate ConfbridgeJoin and ConfbridgeLeave events for the entering and departing channels. Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1	6 years ago
Sean Bright	c70d874f7d	pjproject: Update to 2.9 release ... Relies on https://github.com/asterisk/third-party/pull/4 Change-Id: Iec9cad42cb4ae109a86a3d4dae61e8bce4424ce3	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
George Joseph	93ccff25c6	Merge "app_attended_transfer: new application AttendedTransfer"	6 years ago
Friendly Automation	9bb9700e07	Merge "app_blind_transfer: new application BlindTransfer"	6 years ago
George Joseph	35565edb3b	Merge "chan_pjsip.c: Check for channel and session to not be NULL in hangup"	6 years ago
Alexei Gradinari	3eaeb3e6c4	app_attended_transfer: new application AttendedTransfer ... AttendedTransfer queues up attended transfer to the given extension. This application can be useful with Custom Dynamic Features. For example to make attended transfer to a predefined number. features.conf ;;; [applicationmap] my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_atxfer TRANSFER_CONTEXT=my_transfer [my_atxfer] exten => s,1,AttendedTransfer(1234567890) same => n,Return() [my_transfer] include => default ;;; This application also can be used to completly redefine Attended transfer feature using dialplan. For example: features.conf ;;; [featuremap] atxfer => *7 [applicationmap] custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_atxfer TRANSFER_CONTEXT=my_transfer [custom_atxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,AttendedTransfer(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b	6 years ago
Friendly Automation	b088a17892	Merge "cdr_pgsql: fix error in connection string"	6 years ago
agupta	d2f7b22640	chan_pjsip.c: Check for channel and session to not be NULL in hangup ... We have seen some rare case of segmentation fault in hangup function and we could notice that channel pointer was NULL. Debug log shows that there is a 200 OK answer and SIP timeout at the same time. It looks that while the SIP session was being destroyed due to timeout call hangup due to answer event lead to race condition and channel is being destroyed from two different places. The check ensures we check it not to be NULL before freeing it. ASTERISK-25371 Change-Id: I19f6566830640625e08f7b87bfe15758ad33a778	6 years ago
Alexei Gradinari	745cbab501	app_blind_transfer: new application BlindTransfer ... BlindTransfer redirects all channels currently bridged to the caller channel to the specified destination. This application can be useful with Custom Dynamic Features. For example to make blind transfer to a predefined number. features.conf ;;; [applicationmap] my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_blindxfer [my_blindxfer] exten => s,1,BlindTransfer(1234567890,default) same => n,Return() ;;; This application also can be used to completly redefine Blind transfer feature using dialplan. For example: features.conf ;;; [featuremap] blindxfer => [applicationmap] custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_blindxfer [custom_blindxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,BlindTransfer(${dest},default) same => n,Return() ;;; Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a	6 years ago
Kirsty Tyerman	bcaa01b024	pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi ... ASTERISK-28234 Reported-by: Kirsty Tyerman Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1	6 years ago
Chris-Savinovich	e61f2af89d	cdr_pgsql: fix error in connection string ... Fixes an error occurring in function pgsql_reconnect() caused when value of hostname is blank. Which in turn will cause the connection string to look like this: "host= port=xx", which creates a sintax error. This fix now checks if the corresponding values for host, port, dbname, and user are blank. Note that since this is a reconnect function the database library will replace any missing value pairs with default ones. ASTERISK-28435 Change-Id: I0a921f99bbd265768be08cd492f04b30855b8423	6 years ago
Friendly Automation	03783cfa1b	Merge "res_fax: fix segfault on inactive "reserved" fax session"	6 years ago
Friendly Automation	5750c97eb9	Merge "app_readexten: new option 'p' to stop reading on '#' key"	6 years ago
George Joseph	216cc44e8e	Merge "pjsip: replace 180 by 183 if SDP negotiation has completed"	6 years ago
Friendly Automation	9435788563	Merge "res_fax: add channel name to CLI 'fax show 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
Ben Ford	9969c77bc2	build: Fix file format in CHANGES-staging. ... One of the change files doesn't conform to the format that the release scripts need in order to parse it. Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c	6 years ago
Guido Falsi	db535439f2	chan_dahdi: add missing include. ... After some definitions have been moved to asterisk/mwi.h the files channels/chan_dahdi.h channels/sig_pri.c are missing this new include. ASTERISK-28427 #close Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91	6 years ago
Alexei Gradinari	408210bd4c	app_readexten: new option 'p' to stop reading on '#' key ... This patch adds the 'p' option. The extension entered will be considered complete when a # is entered. Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1	6 years ago
Friendly Automation	9b0a21c402	Merge "res_rtp_asterisk: timestamp should be unsigned instead of signed int"	6 years ago
Friendly Automation	3d4cc9c30d	Merge "pjproject-bundled: Add upstream timer fixes"	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
Matt Jordan	54f7f7dc20	pjproject/Makefile: Updates for Darwin compatible builds ... This patch fixes three compatibility issues for Darwin compatible builds: (1) Use BSD compatible command line option for sed For some versions of BSD sed, the -r command line option is unknown. Both GNU and BSD sed support the -E command line option for enabling extended regular expressions; as such, this patch replaces the -r option with -E. (2) Look for '_' in pjproject generated symbols In Darwin comaptible systems, the symbols generated for pjproject may be prefixed with an '_'. When exporting these to a symbol file, the invocation to sed has to optionally look for a prefix of said '_' character. (3) Use -all_load/-noall_load when linking The flags -whole-archive/-no-whole-archive are not supported by the linker, and must instead be replaced with -all_load/-noall_load. Change-Id: I58121756de6a0560a6e49ca9d6bf9566a333cde3	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
Joshua Colp	3853fab3f5	pjproject-bundled: Add upstream timer fixes ... Fixed #2191: - Stricter double timer entry scheduling prevention. - Integrate group lock in SIP transport, e.g: for add/dec ref, for timer scheduling. ASTERISK-28161 Reported-by: Ross Beer Change-Id: I2e09aa66de0dda9414d8a8259a649c4d2d96a9f5	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	6 years ago
Alexei Gradinari	466a17964f	pjsip: replace 180 by 183 if SDP negotiation has completed ... The caller endpoint hears dead silence if a callee replies 180 (without SDP) and the caller already received 183 (with SDP). It happens because Asterisk sends 180 (WITH SDP) to the caller, there are not incoming RTP packets from the callee and Asterisk does not generate inband ringing, so there are not any outgoing RTP packets to the caller. This patch replaces 180 by 183 if SDP negotiation has completed, as if the caller endpoint is configured with "inband_progress=yes". In this case Asterisk will generate inband ringing untill Asterisk receive incoming RTP packets from the callee. ASTERISK-27994 #close Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73	6 years ago
Friendly Automation	0575a8923b	Merge "res_rtp_asterisk: Fix sequence number cycling and packet loss count."	6 years ago
Friendly Automation	e222dc71d5	Merge "conversions.c: Add conversions for largest max sized integer"	6 years ago
Friendly Automation	9ca99ede61	Merge "Fixes for GCC 9"	6 years ago
Friendly Automation	9db0f08901	Merge "build: Pass --fno-partial-inlining to third-party when appropriate"	6 years ago
Friendly Automation	c0640a3033	Merge "pjsip_options.c: Allow immediate qualifies for new contacts."	6 years ago
George Joseph	c5c953c1f1	Fixes for GCC 9 ... Various fixes for issues caught by gcc 9. Mostly snprintf trying to copy to a buffer potentially too small. ASTERISK-28412 Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e	6 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	6 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	6 years ago
Kevin Harwell	def6bbc96b	conversions.c: Add conversions for largest max sized integer ... Added a conversion for umax (largest maximum sized integer allowed). Adjusted the other current conversion functions (uint and ulong) to be derivatives of the umax conversion since they are simply subsets of umax. Also made the negative check move the pointer on spaces since strtoumax does it anyways. Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08	6 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	6 years ago
George Joseph	089581f20a	build: Pass --fno-partial-inlining to third-party when appropriate ... When the gcc version is >= 8.2.1, we were already setting the --fno-partial-inlining flag for Asterisk source files to get around a gcc bug but we weren't passing the flag down to the bundled builds of pjproject and jansson. ASTERISK-28392 Change-Id: I99ede9bc35408ecd096f7d5369e8192d3dc75704	6 years ago
Friendly Automation	fca9f0a4ad	Merge "app_confbridge: Add "all" variants of REMB behavior."	6 years ago
Joshua Colp	8c1680fce4	Merge "stasis: Only place stasis created and dialed channels into dial bridge."	6 years ago
Friendly Automation	a6d9c4f11a	Merge "stasis: Call callbacks when imparting fails"	6 years ago