4388 Commits (18c8dda84310d149b489b03663e94de542f91157)

Author	SHA1	Message	Date
Jenkins2	3418bd94ff	Merge "res_smdi: Fix shutdown ref." into 15	8 years ago
Jenkins2	f06f311497	Merge "res_rtp_asterisk.c: Disable packet flood detection for video streams." into 15	8 years ago
Jenkins2	9f75ecb153	Merge "res_hep: hepv3_is_loaded() should check if we are enabled" into 15	8 years ago
Joshua Colp	7d3d75b6dc	Merge "res_clialiases: Fix completion pass-through." into 15	8 years ago
Jenkins2	b258f56ba3	Merge "coverity: Fix warnings in res_smdi" into 15	8 years ago
Jenkins2	d4b3c9cec1	Merge "res_musiconhold: Start playlist after initial announcement" into 15	8 years ago
Corey Farrell	044ed1c5d8	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	8 years ago
Sean Bright	1ae46df1ef	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	8 years ago
Corey Farrell	e41bed3713	res_clialiases: Fix completion pass-through. ... Never ignore contents of line when generating completion options. Change-Id: I74389efdfea154019d3b56a9f381610614c044c8	8 years ago
Jenkins2	cff0f36686	Merge "res_pjsip_session: Reinvite using active stream topology if none requested." into 15	8 years ago
Richard Mudgett	39c8d566ad	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	9d63832e28	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	4d4f43f2a0	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	2ef67d2d31	coverity: Fix warnings in res_smdi ... ASTERISK-19657 #close Reported by: Matt Jordan III, Esq. Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a	8 years ago
Joshua Colp	42d797e703	Merge "res_pjsip: Assign support levels to a few modules" into 15	8 years ago
Jenkins2	144d523001	Merge "pjsip_options: contacts sometimes not being updated on reload" into 15	8 years ago
Jenkins2	898889d3f1	Merge "pjsip_options: dynamic contact's fields not updated on reload" into 15	8 years ago
Jenkins2	8986fcfa7a	Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)" into 15	8 years ago
Joshua Colp	6b3bbba6dd	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
Joshua Colp	272594a70a	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
Richard Mudgett	c0822c32ac	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	02bd461c9e	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	bb4fdd3890	res_pjsip: Assign support levels to a few modules ... Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d	8 years ago
Jenkins2	5fb14aab28	Merge "astdb: Improve prefix searches in astdb" into 15	8 years ago
Jenkins2	c421d786d4	Merge "res_stasis and res_speech: Fix load order." into 15	8 years ago
Jenkins2	75af0af465	Merge "utils: Add convenience function for setting fd flags" into 15	8 years ago
Jenkins2	dacbb0f442	Merge "pjsip: Improve CLI completion performance" into 15	8 years ago
Sean Bright	a419bf0b1e	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	d9843c84ef	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
Sean Bright	9b705dee74	pjsip_configuration: Add correct file header ... Change-Id: I25348c386a222bb704aff07f54375108a6402906	8 years ago
Sean Bright	f9c6f69205	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	95a90bdca1	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	6d8d754258	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
Kevin Harwell	845a0d8de3	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	8 years ago
Jenkins2	47124e97fe	Merge "res_rtp_asterisk.c: Increase strictrtp learning timeout time." into 15	8 years ago
Richard Mudgett	04f7a58f67	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	8 years ago
Richard Mudgett	8543582bbd	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	8 years ago
Joshua Colp	ff8e0f78ef	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	8 years ago
Joshua Colp	f78360b371	Merge "res_ari: Fix inverted test giving wrong error message." into 15	8 years ago
Joshua Colp	61158235d2	Merge "res_rtp_asterisk.c: Fix rtp source address learning for broken clients" into 15	8 years ago
Richard Mudgett	338e8c0f0f	res_ari: Fix inverted test giving wrong error message. ... The patch for ASTERISK_24560 inverted a test checking if the bridge name is being updated to a different name. * Fix the test to return "Changing bridge name is not implemented" when someone attempts to change the bridge name. ASTERISK-27445 Change-Id: I4b70bf08b0e02e016108b077ff75b345dec12fc9	8 years ago
Joshua Colp	5d1de66915	Merge "res_parking: Make load_pri explicit." into 15	8 years ago
Joshua Colp	5582566c87	Merge "res_mwi_external_ami: Remove incorrect load priority." into 15	8 years ago
Joshua Colp	67c6f97955	Merge "Loader: Remove unneeded load_pri declarations." into 15	8 years ago
Alexander Traud	73a9b88110	res_rtp_asterisk: ICE server-reflexive candidates (srflx) with Dual-Stack. ... Previously, Asterisk sent srflx only when configured exclusively for IPv4. Now, srflx is gathered and sent via SDP, even when Asterisk is enabled for Dual Stack (IPv4+IPv6) and an IPv4 interface is available/used. ASTERISK-27437 Change-Id: Ie07d8e2bfa7b6fe06fcdc73d390a7a9a4d8c0bc1	8 years ago
Corey Farrell	c784dd2b96	res_parking: Make load_pri explicit. ... res_parking has an implicit load_pri of 0 meaining it's one of the very first modules loaded after modules with global symbols. Set it explicitly in the AST_MODULE_INFO block. Change-Id: I297b6fb3ff6993ec004e667b22a74f5925906259	8 years ago
Joshua Colp	a881c52c78	Merge "res_pjsip: Use sorcery prefix operation for contact lookup" into 15	8 years ago
Corey Farrell	e8c9cb6c40	res_mwi_external_ami: Remove incorrect load priority. ... res_mwi_external_ami specified AST_MODFLAG_LOAD_ORDER but didn't set load_pri, resulting in an actual load priority of 0. This module only provides AMI actions so it has no reason to load early. Change-Id: I82987fcf10d3ea42716b2f9df915b16687fd5839	8 years ago
Corey Farrell	0e2385581b	Loader: Remove unneeded load_pri declarations. ... Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT. Change-Id: I0123258eafce324249433a69df15a85cc16e509f	8 years ago
Joshua Colp	8090aaf823	Merge "res_fax: Remove checks for unsigned values being >= 0." into 15	8 years ago