summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2017-12-18app_transfer: Remove LOCAL from documentation.Joshua Colp
The Local channel has never supported app_transfer from what I can see so remove it from the documentation. ASTERISK-25649 Change-Id: Icbcfe297f6f866285a26b3e9fd5c6d00fa22e0e9
2017-12-15Merge "res_smdi: Fix shutdown ref." into 15Jenkins2
2017-12-15Merge "res_rtp_asterisk.c: Disable packet flood detection for video ↵Jenkins2
streams." into 15
2017-12-15Merge "res_hep: hepv3_is_loaded() should check if we are enabled" into 15Jenkins2
2017-12-15Merge "res_clialiases: Fix completion pass-through." into 15Joshua Colp
2017-12-15Merge "coverity: Fix warnings in res_smdi" into 15Jenkins2
2017-12-15Merge "res_musiconhold: Start playlist after initial announcement" into 15Jenkins2
2017-12-15Merge "app_queue: Fix extension state subscriptions removed on dialplan ↵George Joseph
reload" into 15
2017-12-15res_smdi: Fix shutdown ref.Corey Farrell
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
2017-12-14res_hep: hepv3_is_loaded() should check if we are enabledSean Bright
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
2017-12-14Merge "pjsip: Ignore state changes from old transactions." into 15Jenkins2
2017-12-14res_clialiases: Fix completion pass-through.Corey Farrell
Never ignore contents of line when generating completion options. Change-Id: I74389efdfea154019d3b56a9f381610614c044c8
2017-12-14Merge "res_pjsip_session: Reinvite using active stream topology if none ↵Jenkins2
requested." into 15
2017-12-14Merge "configs: Comment out and change IP of iax.conf [demo]" into 15Jenkins2
2017-12-14res_rtp_asterisk.c: Disable packet flood detection for video streams.Richard Mudgett
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
2017-12-14res_pjsip_sdp_rtp: Add NULL check in add_crypto_to_streamGeorge Joseph
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
2017-12-14res_musiconhold: Start playlist after initial announcementSean Bright
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
2017-12-14coverity: Fix warnings in res_smdiSean Bright
ASTERISK-19657 #close Reported by: Matt Jordan III, Esq. Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a
2017-12-14configs: Comment out and change IP of iax.conf [demo]Sean Bright
This no longer appears to exist, so no sense in causing confusion. ASTERISK-27175 #close Reported by: Tzafrir Cohen Change-Id: Idde967924c69f6a741dc9a5ab7dacb44d22cf100
2017-12-14Merge "menuselect: Tweak check for recently run configure." into 15Joshua Colp
2017-12-14Merge "README: Remove outdated references to tex docs" into 15Joshua Colp
2017-12-13Merge "res_pjsip: Assign support levels to a few modules" into 15Joshua Colp
2017-12-13Merge "pjsip_options: contacts sometimes not being updated on reload" into 15Jenkins2
2017-12-13README: Remove outdated references to tex docsGeorge Joseph
Added links to the wiki to replace references to outdated tex docs. ASTERISK-27430 Reported by: Corey Farrell Change-Id: I5007e732b30bc7b63d124c530ae8857c89991209
2017-12-13Merge "pjsip_options: dynamic contact's fields not updated on reload" into 15Jenkins2
2017-12-13Merge "CLI: Fix 'core show sysinfo' function ordering." into 15Jenkins2
2017-12-13Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)" into 15Jenkins2
2017-12-13Merge "AST-2017-012: Place single RTCP report block at beginning of report." ↵George Joseph
into 15
2017-12-13AST-2017-012: Place single RTCP report block at beginning of report.Joshua Colp
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
2017-12-13Merge "chan_sip: Don't crash in Dial on invalid destination" into 15Joshua Colp
2017-12-13res_pjsip_session: Reinvite using active stream topology if none requested.Joshua Colp
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
2017-12-13Merge "chan_sip: Don't send trailing \0 on keep alive packets" into 15Jenkins2
2017-12-13pjsip: Ignore state changes from old transactions.Joshua Colp
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
2017-12-12app_queue: Fix extension state subscriptions removed on dialplan reloadIvan Poddubny
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
2017-12-12chan_sip: Don't send trailing \0 on keep alive packetsSean Bright
This is a partial fix for ASTERISK~25817 but does not address the comments regarding RFC 5626. Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420
2017-12-12chan_sip: Don't crash in Dial on invalid destinationSean Bright
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
2017-12-12menuselect: Tweak check for recently run configure.Corey Farrell
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
2017-12-12chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)Richard Mudgett
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
2017-12-12res_pjsip: Add TLSv1.1 and TLSv1.2 supportSean Bright
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
2017-12-12res_pjsip: Assign support levels to a few modulesSean Bright
Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d
2017-12-11CLI: Fix 'core show sysinfo' function ordering.Corey Farrell
Handle CLI initialization before any processing occurs. Change-Id: I598b911d2e409214bbdfd0ba0882be1d602d221c
2017-12-11stasis_channels.c: Don't set channel snapshot caller_dnid twice.Richard Mudgett
Change-Id: Ib8d45bbdfbda81e65045f6dff874d189b74e5471
2017-12-11Merge "codec_opus: Make libcurl a dependency in menuselect" into 15George Joseph
2017-12-11Merge "astdb: Improve prefix searches in astdb" into 15Jenkins2
2017-12-11Merge "loader: Refactor resource_name_match." into 15Jenkins2
2017-12-11Merge "res_stasis and res_speech: Fix load order." into 15Jenkins2
2017-12-11Merge "utils: Add convenience function for setting fd flags" into 15Jenkins2
2017-12-11codec_opus: Make libcurl a dependency in menuselectSean Bright
ASTERISK-27475 #close Change-Id: If7384bc6ed002ef140dec69798d14c52b7cfd800
2017-12-11Merge "pjsip: Improve CLI completion performance" into 15Jenkins2
2017-12-11Merge "CDR: Fix deadlock setting some CDR values." into 15Jenkins2
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-26 17:37:50 +0000