asterisk

Commit Graph

Author	SHA1	Message	Date
Corey Farrell	a8aa209901	res_clialiases: Fix completion pass-through. Never ignore contents of line when generating completion options. Change-Id: I74389efdfea154019d3b56a9f381610614c044c8	8 years ago
Jenkins2	a33207a91f	Merge "res_pjsip_session: Reinvite using active stream topology if none requested."	8 years ago
Richard Mudgett	98f7e9251f	res_rtp_asterisk.c: Disable packet flood detection for video streams. We should not do flood detection on video RTP streams. Video RTP streams are very bursty by nature. They send out a burst of packets to update the video frame then wait for the next video frame update. Really only audio streams can be checked for flooding. The others are either bursty or don't have a set rate. * Added code to selectively disable packet flood detection for video RTP streams. ASTERISK-27440 Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70	8 years ago
George Joseph	283d2df680	res_pjsip_sdp_rtp: Add NULL check in add_crypto_to_stream add_crypto_to_stream wasn't checking for a NULL session->inv_session->neg before calling pjmedia_sdp_neg_get_state. This was causing a crash if the negotiation hadn't already been completed and asterisk was compiled with --enable-dev-mode. Change-Id: I57c6229954a38145da9810fc18657bfcc4d9d0c9	8 years ago
Sean Bright	c387beb456	res_musiconhold: Start playlist after initial announcement Reset the samples counter to zero when we are done playing an announcement so that we don't skip into the middle of the first file in the playlist. Also add the selected annoucement to the output of 'moh show classes.' ASTERISK-24329 #close Reported by: Thomas Frederiksen Change-Id: I2a5f986a31279c981592f49391409ebf38d6f6d0	8 years ago
Sean Bright	7a8a187a56	coverity: Fix warnings in res_smdi ASTERISK-19657 #close Reported by: Matt Jordan III, Esq. Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a	8 years ago
Sean Bright	dac5e3a0df	configs: Comment out and change IP of iax.conf [demo] This no longer appears to exist, so no sense in causing confusion. ASTERISK-27175 #close Reported by: Tzafrir Cohen Change-Id: Idde967924c69f6a741dc9a5ab7dacb44d22cf100	8 years ago
Joshua Colp	ec984870b0	Merge "menuselect: Tweak check for recently run configure."	8 years ago
Joshua Colp	49f2ff37bd	Merge "README: Remove outdated references to tex docs"	8 years ago
Joshua Colp	7a837ece97	Merge "Add new AMI action for app_voicemail"	8 years ago
Joshua Colp	e411b7d112	Merge "chan_sip: 3PCC patch for AMI "SIPnotify""	8 years ago
Kevin Harwell	30954337a0	Merge "pjsip_options: contacts sometimes not being updated on reload"	8 years ago
Jenkins2	588be919cb	Merge "res_pjsip: Assign support levels to a few modules"	8 years ago
George Joseph	a51bfe5a79	README: Remove outdated references to tex docs Added links to the wiki to replace references to outdated tex docs. ASTERISK-27430 Reported by: Corey Farrell Change-Id: I5007e732b30bc7b63d124c530ae8857c89991209	8 years ago
Jenkins2	8a281776d5	Merge "pjsip_options: dynamic contact's fields not updated on reload"	8 years ago
George Joseph	afb6e85980	Merge "CLI: Fix 'core show sysinfo' function ordering."	8 years ago
Joshua Colp	c50905756b	Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)"	8 years ago
Corey Farrell	5f6a3c4399	CLI: Remove special handling of 'core set verbose' from rasterisk. rasterisk does not need to handle setting verbose levels locally, it should just tell the daemon what it wants and print what it is given. Just max out the verbose level on the local client so all filtering happens on the daemon. ASTERISK-20281 #close Change-Id: Ia305f75f1fc424a9169bfa30ef70d626ace2c8a8	8 years ago
pchero	daa3a3009a	Add new AMI action for app_voicemail Currently, to figure out specified voicemail's status, there's only one way to do it, which is use a VoicemailUserEntry AMI message. But it consumed it too much resource(it check everything). So, added new AMI action. ASTERISK-27470 Change-Id: Ie4eba1424a142e5fbd1d9fb1821a3fc1a1e238b7	8 years ago
Joshua Colp	62f2860c39	AST-2017-012: Place single RTCP report block at beginning of report. When the RTCP code was transitioned over to Stasis a code change was made to keep track of how many reports are present. This count controlled where report blocks were placed in the RTCP report. If a compound RTCP packet was received this logic would incorrectly place a report block in the wrong location resulting in a write to an invalid location. This change removes this counting logic and always places the report block at the first position. If in the future multiple reports are supported the logic can be extended but for now keeping a count serves no purpose. ASTERISK-27382 ASTERISK-27429 Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116	8 years ago
Jenkins2	58514c7442	Merge "chan_sip: Don't crash in Dial on invalid destination"	8 years ago
Joshua Colp	3370cd21df	res_pjsip_session: Reinvite using active stream topology if none requested. When a connected line update is sent to an endpoint we do not request a specific stream topology to be used. Previously this resulted in the configured stream topology being used which may actually differ from the currently negotiated topology. PJSIP is helpful in this regard in that it will fill in any missing streams with removed ones. This results in our own state not matching the SDP, though, and we do not apply the negotiated SDP. This change tweaks the code to use the actively negotiated stream topology if it is present with a fallback to the configured one. This results in the SDP and the state having matching information and the world is happy. ASTERISK*27397 Change-Id: I7a57117f0183479e6884b7bf3a53bb8c7464f604	8 years ago
Jenkins2	e7dccbe708	Merge "chan_sip: Don't send trailing \0 on keep alive packets"	8 years ago
Joshua Colp	0b532367bd	pjsip: Ignore state changes from old transactions. When we fail over to a new target we create a new transaction and it becomes the current INVITE transaction. This does not prevent the previous transaction from raising state changes and causing the session to be prematurely disconnected if a transport error occurs immediately. This change backports a fix from PJSIP that eliminates the incorrect state change and reduces when they would be raised in the first place. ASTERISK-27408 Change-Id: Id22d087591782eee31311753d11e7eca4b95ef34	8 years ago
Yasuhiko Kamata	cb249b2419	chan_sip: 3PCC patch for AMI "SIPnotify" A patch for sending in-dialog SIP NOTIFY message with "SIPnotify" AMI action. ASTERISK-27461 Change-Id: I5797ded4752acd966db6b13971284db684cc5ab4	8 years ago
Ivan Poddubny	c7f94e570e	app_queue: Fix extension state subscriptions removed on dialplan reload The approach with having a single global subscription to all extension state changes has one issue: dynamically created hints don't have any watchers and are therefore garbage collected on the first dialplan reload. This change creates a state subscription for every queue member with a hint as state_interface, thus increasing the count of watches for hints, so they are not destroyed prematurely anymore. There are 2 side effects: 1. The state change callback in app_queue is not executed when there are no members referring to the extension. 2. The callback is called multiple times for the same hint if it's associated with more than one queue member. Reported by: Steven T. Wheeler ASTERISK-18411 #close Change-Id: I4956af2136ea2a7f110ac9272eae5f6e676d8f89	8 years ago
Sean Bright	0c9cc7e975	chan_sip: Don't send trailing \0 on keep alive packets This is a partial fix for ASTERISK~25817 but does not address the comments regarding RFC 5626. Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420	8 years ago
Sean Bright	5039b5741c	chan_sip: Don't crash in Dial on invalid destination Stripping the DNID in a SIP dial string can result in attempting to call the argument parsing macros on an empty string, causing a crash. ASTERISK-26131 #close Reported by: Dwayne Hubbard Patches: dw-asterisk-master-dnid-crash.patch (license #6257) patch uploaded by Dwayne Hubbard Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e	8 years ago
Corey Farrell	6a67828b46	menuselect: Tweak check for recently run configure. Recently menuselect has randomly produced an error stating that configure was just run and make had to be restarted. I believe this is due to an incorrect menuselect/Makefile rule. The original rule produced an error if makeopts or autoconfig.h were older than makeopts.in or autoconfig.h.in. I believe this can create an issue if makeopts is older than autoconfig.h.in or if autoconfig.h is older than makeopts.in. The new rules compare files independently. Change-Id: Ibca155035fa1392c95e33cbf25f257902abba17b	8 years ago
Richard Mudgett	22810fc635	chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri) This patch does three things associated with the initial incoming INVITE request URI. 1) Add access to the full initial incoming INVITE request URI. 2) We were not setting DNID on incoming PJSIP channels. The DNID is the user portion of the initial incoming INVITE Request-URI. The value is accessed by reading CALLERID(dnid). 3) Fix CHANNEL(pjsip,target_uri) documentation. * The initial incoming INVITE request URI is now available using CHANNEL(pjsip,request_uri). * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the initial incoming INVITE request URI user portion. * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is the contact URI. * Refactored print_escaped_uri() out of channel_read_pjsip() to handle pjsip_uri_print() error condition when the buffer is too small. ASTERISK-27478 Change-Id: I512e60d1f162395c946451becb37af3333337b33	8 years ago
Sean Bright	ec1f4bf48d	res_pjsip: Add TLSv1.1 and TLSv1.2 support Support for these protocols was added in the same commit as the 'proto' field, so we can safely use the same ./configure check. For reference: https://trac.pjsip.org/repos/changeset/4968 Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac	8 years ago
Sean Bright	0b9d2135a9	res_pjsip: Assign support levels to a few modules Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d	8 years ago
Corey Farrell	c01ba7437e	CLI: Fix 'core show sysinfo' function ordering. Handle CLI initialization before any processing occurs. Change-Id: I598b911d2e409214bbdfd0ba0882be1d602d221c	8 years ago
Kevin Harwell	b088cddc03	pjsip_options: wrongly applied "UNKNOWN" status A couple of places were setting the status to "UNKNOWN" when qualifies were being disabled. Instead this should be set to the "CREATED" status that represents when a contact is given (uri available), but the qualify frequency is set to zero so we don't know the status. This patch updates the relevant places with "CREATED". It also updates the "CREATED" status description (value shown in CLI/AMI/ARI output) to a value of "NonQualified"/"NonQual" as this description is hopefully less confusing. ASTERISK-27467 Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89	8 years ago
Richard Mudgett	c2ec82bf36	stasis_channels.c: Don't set channel snapshot caller_dnid twice. Change-Id: Ib8d45bbdfbda81e65045f6dff874d189b74e5471	8 years ago
Jenkins2	38dcdf2d68	Merge "codec_opus: Make libcurl a dependency in menuselect"	8 years ago
Jenkins2	b7e79d7baf	Merge "astdb: Improve prefix searches in astdb"	8 years ago
Jenkins2	f81a884033	Merge "loader: Refactor resource_name_match."	8 years ago
Joshua Colp	5d43a4d4ff	Merge "res_stasis and res_speech: Fix load order."	8 years ago
Jenkins2	710b3a29c3	Merge "utils: Add convenience function for setting fd flags"	8 years ago
Sean Bright	00578fae0a	codec_opus: Make libcurl a dependency in menuselect ASTERISK-27475 #close Change-Id: If7384bc6ed002ef140dec69798d14c52b7cfd800	8 years ago
Jenkins2	cd741136c5	Merge "pjsip: Improve CLI completion performance"	8 years ago
Jenkins2	6d25b26933	Merge "CDR: Fix deadlock setting some CDR values."	8 years ago
Sean Bright	521f741b04	pjsip: Improve CLI completion performance Use the new ast_cli_completion_add() function to improve completion performance for commands like 'pjsip show endpoint.' Change-Id: I76d802294d2ac1766110dc75f7d117c8541ce348	8 years ago
Sean Bright	9a9edc6c9e	astdb: Improve prefix searches in astdb Using the LIKE operator requires a full table scan of 'astdb', whereas a comparison operation is able to use the primary key index. This patch adds a new function to the AstDB API for quick prefix matches and updates res_sorcery_astdb to utilize it. This showed substantial performance improvement in my test environment. Related to ASTERISK~26806, but does not completely resolve it. Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1	8 years ago
Corey Farrell	d2e87b8e14	loader: Refactor resource_name_match. Optimize resource_name_match. This change eliminates use of ast_strdupa, instead verifying that both basename's are the same length, then using strncasecmp. Change-Id: I477275c0e954c99d74be5abfc8bb6545b04e5a3d	8 years ago
Sean Bright	dbb376f166	pjsip_configuration: Add correct file header Change-Id: I25348c386a222bb704aff07f54375108a6402906	8 years ago
Sean Bright	2ffe52a116	utils: Add convenience function for setting fd flags There are many places in the code base where we ignore the return value of fcntl() when getting/setting file descriptior flags. This patch introduces a convenience function that allows setting or clearing file descriptor flags and will also log an error on failure for later analysis. Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7	8 years ago
Corey Farrell	e2dbc26376	res_stasis and res_speech: Fix load order. res_stasis was missing AST_MODFLAG_LOAD_ORDER. Set res_stasis and res_speech to start at (AST_MODPRI_APP_DEPEND - 1) so they are ready for dependent modules. Change-Id: I27f4f3810a95b6be8a5bfbf62be2ace6bfab6ff3	8 years ago
Kevin Harwell	0e4d31eb9c	pjsip_options: contacts sometimes not being updated on reload For both dynamic and static contacts it was possible that potential AOR changes were not being applied to all contacts. This was because the qualify and schedule code was only retrieving AOR's, and contacts with frequencies greater than zero. For instance the following could happen: and AOR/contact has a frequency of 5, it then gets set to 0, and then a reload occurs. All scheduled OPTIONS are stopped, a list of AOR's is retrieved with frequency > 0, but none are selected since in this scenario all are 0. The contact for the one previously set to 5 though does not get updated, so it's status remains "AVAILABLE". This patch makes it so all contacts (static and dynamic) are selected, and appropriately updated if need be. ASTERISK-27467 #close Change-Id: I7a920170f89c683af9505d4723a44fc6841decdb	8 years ago

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