asterisk

Commit Graph

Author	SHA1	Message	Date
Sungtae Kim	ffbf5be116	res_pjsip: Add AMI action 'PJSIPShowAors' Add an AMI action which provides information on all configured AORs. ASTERISK-27537 Change-Id: If8b990a00909e5b6c0f04a3b8dccd9903dc445eb	9 years ago
Kevin Harwell	553306548c	AST-2017-014: res_pjsip - Missing contact header can cause crash Those SIP messages that create dialogs require a contact header to be present. If the contact header was missing from the message it could cause Asterisk to crash. This patch checks to make sure SIP messages that create a dialog contain the contact header. If the message does not and it is required Asterisk now returns a "400 Missing Contact header" response. Also added NULL checks when retrieving the contact header that were missing as a "just in case". ASTERISK-27480 #close Change-Id: I1810db87683fc637a9e3e1384a746037fec20afe	9 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	9 years ago
Corey Farrell	1b80ffa495	Fix Common Typo's. Fix instances of: * Retreive * Recieve * other then * different then * Repeated words ("the the", "an an", "and and", etc). * othterwise, teh ASTERISK-24198 #close Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31	9 years ago
Jenkins2	78fb99e5a3	Merge "res_rtp_asterisk: Avoid close the rtp/rtcp fd twice."	9 years ago
Corey Farrell	b3e839debd	Remove constant conditionals (dead-code). Some variables are set and never changed, making them constant. This means that code in the 'false' block of the conditional is unreachable. In chan_skinny and res_config_ldap I used preprocessor directive `#if 0` as I'm unsure if the unreachable code could be enabled in the future. Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059	9 years ago
Aaron An	81474dfb23	res_rtp_asterisk: Avoid close the rtp/rtcp fd twice. When RTCP-MUX enabled. rtp->s is the same as rtcp->s, check this before close the file descriptor. Close the FD twice will hangs the asterisk under heavy load. ASTERISK-27299 #close Reported-by: Aaron An Tested-by: AaronAn Change-Id: I870a072d73fd207463ac116ef97100addbc0820a	9 years ago
Joshua Colp	5224fd3ab4	Merge changes from topic 'faster-aco' * changes: aco: Minimize use of regex. aco: Create ways to minimize use of regex.	9 years ago
Jenkins2	eb23919e69	Merge "res_smdi: Fix shutdown ref."	9 years ago
Jenkins2	c9bcd888a2	Merge "res_rtp_asterisk.c: Disable packet flood detection for video streams."	9 years ago
Jenkins2	6a0505eee0	Merge "res_hep: hepv3_is_loaded() should check if we are enabled"	9 years ago
Joshua Colp	73bd9d6488	Merge "res_clialiases: Fix completion pass-through."	9 years ago
Jenkins2	26e8de9453	Merge "coverity: Fix warnings in res_smdi"	9 years ago
Jenkins2	bcb4e6e608	Merge "res_musiconhold: Start playlist after initial announcement"	9 years ago
Jenkins2	dff0415b1e	Merge "pjsip_options: wrongly applied "UNKNOWN" status"	9 years ago
Corey Farrell	bf2d35931d	aco: Minimize use of regex. Remove nearly all use of regex from ACO users. Still remaining: * app_confbridge has a legitamate use of option name regex. * ast_sorcery_object_fields_register is implemented with regex, all callers use simple prefix based regex. I haven't decided the best way to fix this in both 13/15 and master. Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b	9 years ago
Corey Farrell	03c25a869f	res_smdi: Fix shutdown ref. When adding shutdown refs for OPTIONAL_API components I accidentally added it to the unload_module function in res_smdi. Move it to load_module. Change-Id: I2b9da38fbc11ef78ea23dbb2df92b684be7f647c	9 years ago
Sean Bright	9755eff46f	res_hep: hepv3_is_loaded() should check if we are enabled res_hep_pjsip.so and res_hep_rtcp.so will still load and do a lot of unnecessary work even if 'enabled' is set to 'no' in hep.conf. Change-Id: I3eddfeea09c6b5bc7c641952ee0ae487fd09b64b	9 years ago
Corey Farrell	a8aa209901	res_clialiases: Fix completion pass-through. Never ignore contents of line when generating completion options. Change-Id: I74389efdfea154019d3b56a9f381610614c044c8	9 years ago
Jenkins2	a33207a91f	Merge "res_pjsip_session: Reinvite using active stream topology if none requested."	9 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	9 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	9 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	9 years ago
Sean Bright	7a8a187a56	coverity: Fix warnings in res_smdi ASTERISK-19657 #close Reported by: Matt Jordan III, Esq. Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a	9 years ago
Kevin Harwell	30954337a0	Merge "pjsip_options: contacts sometimes not being updated on reload"	9 years ago
Jenkins2	588be919cb	Merge "res_pjsip: Assign support levels to a few modules"	9 years ago
Jenkins2	8a281776d5	Merge "pjsip_options: dynamic contact's fields not updated on reload"	9 years ago
Joshua Colp	c50905756b	Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)"	9 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	9 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	9 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	9 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	9 years ago
Sean Bright	0b9d2135a9	res_pjsip: Assign support levels to a few modules Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d	9 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	9 years ago
Jenkins2	b7e79d7baf	Merge "astdb: Improve prefix searches in astdb"	9 years ago
Joshua Colp	5d43a4d4ff	Merge "res_stasis and res_speech: Fix load order."	9 years ago
Jenkins2	710b3a29c3	Merge "utils: Add convenience function for setting fd flags"	9 years ago
Jenkins2	cd741136c5	Merge "pjsip: Improve CLI completion performance"	9 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	9 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	9 years ago
Sean Bright	dbb376f166	pjsip_configuration: Add correct file header Change-Id: I25348c386a222bb704aff07f54375108a6402906	9 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	9 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	9 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	9 years ago
Kevin Harwell	bd2218ce63	pjsip_options: dynamic contact's fields not updated on reload Dynamic contacts were not being properly updated on reload. As a matter of fact any changes to the AOR that a dynamic contact was associated with were not being applied. On reload, this patch makes it so for each dynamic contact, the associated AOR is now retrieved and the AOR's fields are applied to the contact. ASTERISK-27467 Change-Id: I8e3165dc6a745218c1c9db837f77fafa0516985d	9 years ago
Joshua Colp	628a0af6de	Merge "res_rtp_asterisk.c: Increase strictrtp learning timeout time."	9 years ago
Richard Mudgett	8536a09b86	security-events: Fix SuccessfulAuth using_password declaration. The SuccessfulAuth using_password field was declared as a pointer to a uint32_t when the field was later read as a uint32_t value. This resulted in unnecessary casts and a non-portable field value reinterpret in main/security_events.c:add_json_object(). i.e., It would work on a 32 bit architecture but not on a 64 bit big endian architecture. Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935	9 years ago
Richard Mudgett	ab63448fa6	res_rtp_asterisk.c: Increase strictrtp learning timeout time. More complicated direct media reinvite negotiations can result in longer delays before direct media flows. The strictrtp learning timeout time was too short. One log showed that the first RTP packet came in just after three seconds. * Increase the strictrtp learning timeout time from 1.5 to 5 seconds. ASTERISK-27453 Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c	9 years ago
Joshua Colp	892df22ccd	res_http_post: Not all versions of gmime have GMIME_MAJOR_VERSION. This change makes the presence of the GMIME_MAJOR_VERSION definition optional, as not all versions of gmime actually define it. ASTERISK-27454 Change-Id: I01d99590045971ed6787899147170a5954077238	9 years ago
Joshua Colp	8bf1a5c46a	Merge "res_ari: Fix inverted test giving wrong error message."	9 years ago

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4405 Commits (ffbf5be116a3035f0ecf97bd13d87bfd64048fb0)