32263 Commits (9e8953518279070fff6d139edd7cfff7f80ef354)

 

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Author	SHA1	Message	Date
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
Francesco Castellano	8438d19b81	chan_sip: Handle invalid SDP answer to T.38 re-invite ... The chan_sip module performs a T.38 re-invite using a single media stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional media stream with no joint codec a crash will occur as the code assumes that at least one joint codec will exist in this scenario. This change removes this assumption. ASTERISK-28465 Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87	6 years ago
Kevin Harwell	c93c579190	app_voicemail: Remove dependency on the stasis cache ... app_voicemail utilized the stasis cache when polling mailboxes for MWI. This caused a memory leak (items were not being appropriately removed from the cache), and subsequent slowdown in system processing. This patch removes the stasis cache dependency, thus alleviating the memory leak. It does this by utilizing the new MWI API that better manages state lifetime. ASTERISK-28443 ASTERISK-27121 Change-Id: Ie89fedaca81ea1fd03d150d9d3a1ef3d53740e46	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	b31ac83900	mwi: Update the MWI core to use stasis_state API ... ** Note ** This patch is meant to be the minimum needed in order for the MWI core to use the now underlying stasis_state module. As such it does not completely remove its reliance on the stasis_cache. Doing so has allowed current consumers to not have to change, and update those code paths for this patch. When time allows, subsequent patches can/will be made to those consumers to take advantage of some of the new MWI API included here. Thus, eventually and ultimately removing MWI dependency on the stasis_cache. ** End Note ** This patch makes it so the MWI core now takes advantage of the new stasis_state API. Consumers of MWI should no longer need to depend upon stasis topic pooling, and the stasis cache directly. Similar functionality and implementation details have now been pushed into the stasis_state module. However, all MWI state should be accessed via the MWI API itself. As such a few new methods, and constructs have been added to the MWI core that facilitate consumer publishing, subscribing, and iterating over MWI state data. * ast_mwi_subscriber * Created via ast_mwi_add_subscriber, a subscriber subscribes to a given mailbox in order to receive updates about the given mailbox. Adding a subscriber will create the underlying topic, and associated state data if those do not already exist for it. The topic, and last known state data is guaranteed to exist for the lifetime of the subscriber. * ast_mwi_publisher * Before publishing to a particular topic a publisher should be created. This can be achieved by using ast_mwi_add_publisher. Publishing to a mailbox should then be done using one of the MWI publish functions. This ensures the message is published to the appropriate topic, and the last known state is maintained. * ast_mwi_observer * Add an observer in order to watch for particular MWI module related events. For instance if a submodule needs to know when a subscription is added to any mailbox an observer can be added to watch for that. * other * Urgent message count is now part of the published MWI state object. Also state can be iterated over using defined callbacks. ASTERISK-28442 Change-Id: I93f935f9090cd5ddff6d4bc80ff90703c05cf776	6 years ago
Kevin Harwell	83c6ebbae8	stasis_state: Make unsubscribes NULL tolerant ... Regular stasis unsubscribes can handle NULL subscription objects. This patch makes it so stasis state unsubscribes handles NULL's as well. ASTERISK-28442 Change-Id: Ic3648e8df043a85b77cff085e9ff10356028e479	6 years ago
Rodrigo Ramírez Norambuena	64a908f897	README.md: Update year ... Change-Id: I746fb94d112c7d797e206bca0fd1e13fcd26bae3	6 years ago
Friendly Automation	99addaff69	Merge "stasis_state: Add new stasis_state module"	6 years ago
Joshua Colp	c080f01d75	Merge "chan_dahdi.c: crash in chan_dahdi"	6 years ago
Chris-Savinovich	0e669712e2	chan_dahdi.c: crash in chan_dahdi ... Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous patch introduced a variable of type unassigned long long which is 64-bits. Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work with 32-bit systems. ASTERISK-28457 Change-Id: I9cef6b5f2d826fc5c93f2f6a1c997c4e3e6c93fe	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
George Joseph	ad720adda7	Merge "app_voicemail.c: Build all three variants for app_voicemail at the same time"	6 years ago
George Joseph	4d0cc7bbc4	Merge "tcptls.c: Add peer hostname and port to some error messages"	6 years ago
Friendly Automation	7034bb1b21	Merge "pjproject_bundled: Add peer information to most SSL/TLS errors"	6 years ago
Kevin Harwell	363bafc29e	stasis_state: Add new stasis_state module ... This new module describes an API that can be thought of as a combination of stasis topic pools, and caching. Except, hopefully done in a more efficient and less memory "leaky" manner. The API defines methods, and data structures for managing, and tracking published message state through stasis. By adding a subscriber or publisher, consumers can more easily track the lifetime of the contained state. For instance, when no more publishers and/or subscribers have need of the topic, and associated state its data is removed from the managed container. * stasis_state_manager * The manager stores and well, manages state data. Each state is an association of a unique stasis topic, and the last known published stasis message on that topic. There is only ever one managed state object per topic. For each topic all messages are forwarded to an "all" topic also maintained by the manager. * stasis_state_subscriber * Topic and state can be created, or referenced within the manager by adding a stasis_state_subscriber. When adding a subscriber if no state currently exists new managed state is immediately created. If managed state already exists then a new subscriber is created referencing that state. The managed state is guaranteed to live throughout the subscriber's lifetime. State is only removed from the manager when no other entities require it. * stasis_state_publisher * Topic and state can be created, or referenced within the manager by also adding a stasis_state_publisher. When adding a publisher if no state currently exists new managed state is created. If managed state already exists then a new publisher is created referencing that state. The managed state is guaranteed to live throughout the publisher's lifetime. State is only removed from the manager when no other entities require it. * stasis_state_observer * Some modules may wish to watch for, and react to managed state events. By registering a state observer, and implementing handlers for the desired callbacks those modules can do so. * other * Callbacks also exist that allow consumers to iterate over all, or some of the managed state. ASTERISK-28442 Change-Id: I7a4a06685a96e511da9f5bd23f9601642d7bd8e5	6 years ago
Chris-Savinovich	6b1f6ea2c4	app_voicemail.c: Build all three variants for app_voicemail at the same time ... Changes made to apps/Makefile to optionally build all three app_voicemail variations at the same time: 1) file (default), 2) odbc, and 3) imap. This functionality was requested by users. modules.conf.sample warns the user to make sure only one voicemail is loaded at a time. Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7	6 years ago
Kevin Harwell	09508e0671	Merge "pjproject: Update to 2.9 release"	6 years ago
George Joseph	c2ffb004aa	tcptls.c: Add peer hostname and port to some error messages ... Where possble, hostname and port has been added to error messages, mostly on the server side. ASTERISK-26006 Reported by: Oleksandr Natalenko Change-Id: Iff4f897277bc36ce8c5b493b71d0a4a7b74e62f0	6 years ago
George Joseph	8b3ee7fe61	pjproject_bundled: Add peer information to most SSL/TLS errors ... Most SSL/TLS error messages coming from pjproject now have either the peer address:port or peer hostname, depending on what was available at the time and code location where the error was generated. ASTERISK-28444 Reported by: Bernhard Schmidt Change-Id: I41770e8a1ea5e96f6e16b236692c4269ce1ba91e	6 years ago
Friendly Automation	2f29c375c1	Merge "res/ari/resource_channels.c: Added hangup reason code for channels"	6 years ago
Kevin Harwell	cfdb567425	Merge "app_amd: issue with silence suppression fixed"	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
George Joseph	ccbc83fed7	Merge "sig_pri: Address gcc9 issues"	6 years ago
Dan Cropp	e52fbae00f	chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS ... Previously, when a Transfer (REFER) was performed, chan_pjsip would set the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not reflect a successful/unsuccessful transfer the way chan_sip did. Added a callback module to process the refer subscription information. Now depends on res_pjsip_pubsub so call transfer progress can be monitored and reported ASTERISK-26968 #close Reported-by: Dan Cropp Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc	6 years ago
George Joseph	c3adb60dac	Merge "CI: New way to determnine libdir"	6 years ago
George Joseph	619a50b7aa	Merge "res_fax: gateway sends T.38 request to both endpoints if V.21 detected"	6 years ago
George Joseph	13e89d372b	sig_pri: Address gcc9 issues ... A few more format truncation issues addressed. Change-Id: I047f373169caaca0eec4889d3c0e5e10f130017a	6 years ago
George Joseph	3a51cdad18	Merge "translate.c do not log WARNING on empty audio frame"	6 years ago
Friendly Automation	0a54b6c26a	Merge "app_confbridge: Attended transfer event fixup"	6 years ago
Nasir Iqbal	29bc7cf6b3	app_amd: issue with silence suppression fixed ... Now AMD algorithm will not ignore AST_FRAME_NULL, As I think using manual wait time instead of `framelength` is enough to fix timeout / TOOLONG issue. ASTERISK-28419 #close Change-Id: I16ea2d6295bc99b975e8c092e5f9fbd9214debdb	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
George Joseph	0ba52ce3cf	CI: New way to determnine libdir ... We were using the presence of /usr/lib64 to determine where shared libraries should be installed. This only existed on Redhat based systems and was safe. If it existed, use it, otherwise use /usr/lib. Unfortunately, Ubuntu 19 decided to create a /usr/lib64 BUT NOT INCLUDE IT IN THE DEFAULT ld.so.conf. So if anything is installed there, it won't work. The new method, just looks for $ID in /etc/os-release and if it's centos or fedora, uses /usr/lib64 and if ubuntu, uses /usr/lib. NOTE: This applies only to the CI scripts. Normal asterisk build and install is not affected. Change-Id: Iad66374b550fd89349bedbbf2b93f8edd195a7c3	6 years ago
Alexei Gradinari	e3866cb714	translate.c do not log WARNING on empty audio frame ... There is WARNING "no samples for ..." on each Playtones. The function ast_playtones_start calls ast_activate_generator, which calls ast_prod. The function ast_prod calls ast_write with empty audio frame. In this case it's spam log. Change-Id: Id4ac309489d9ff281bad02abdef341cecdede660	6 years ago
George Joseph	92d4ec2906	chan_dahdi: Address gcc9 issues ... Fixed format-truncation issues in chan_dahdi.c and sig_analog.c. Since they're related to fields provided by dahdi-tools we can't change the buffer sizes so we're just checking the return from snprintf and printing an errior if we overflow. Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5	6 years ago
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	7 years ago
Friendly Automation	03783cfa1b	Merge "res_fax: fix segfault on inactive "reserved" fax session"	7 years ago
Friendly Automation	5750c97eb9	Merge "app_readexten: new option 'p' to stop reading on '#' key"	7 years ago
George Joseph	216cc44e8e	Merge "pjsip: replace 180 by 183 if SDP negotiation has completed"	7 years ago
Friendly Automation	9435788563	Merge "res_fax: add channel name to CLI 'fax show session'"	7 years ago