32557 Commits (1b01a6abfd1f323c9043fc9c65f4ab038296f77c)

 

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Author	SHA1	Message	Date
Walter Doekes	b455d57f7d	app_queue: Refactor odd placement of if's around say_position ... Change-Id: Icba97905e331812f129e5966e91a59b104c7a748	5 years ago
Kevin Harwell	d77a89cb9e	format_cap: make function parameters 'const' ... There were a couple places where the format cap function parameter was not 'const' when it should have been. This patch makes them 'const'. Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e	5 years ago
Jaco Kroon	77d4923307	addons/res_config_mysql: silense warnings about printf format errors. ... Warnings without this: res_config_mysql.c: In function 'update2_mysql': res_config_mysql.c:741:15: warning: format '%llu' expects argument of type 'long long unsigned int', but argument 6 has type 'my_ulonglong' {aka 'long unsigned int'} [-Wformat=] ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n", numrows, tablename); (reformatted for readability within line-wrap) Change-Id: I2af4d419a37c1a7eeee750cf9ae4a9a2b3a37fd3	5 years ago
George Joseph	fd31ef67b9	Merge "tcptls.c: Log more informative OpenSSL errors" into 17	5 years ago
George Joseph	e4f5142e23	Merge "bridging: Add better support for adding/removing streams." into 17	5 years ago
George Joseph	3e4752165d	Merge "ast_tls_cert: Allow private key size to be set on command line" into 17	5 years ago
George Joseph	e300d57134	Merge "app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used" into 17	5 years ago
George Joseph	f1896e9279	Merge "RTP/ICE: Send on first valid pair." into 17	5 years ago
Joshua C. Colp	bc3c095e9f	pjsip: Update ACLs on named ACL changes. ... This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b	5 years ago
Sean Bright	9ef514ae98	tcptls.c: Log more informative OpenSSL errors ... Dump OpenSSL's error stack to the error log when things fail. ASTERISK-28750 #close Reported by: Martin Zeh Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a	5 years ago
Sean Bright	cf26ce5d4f	ast_tls_cert: Allow private key size to be set on command line ... The default size in release branches will be 1024 but we'll use 2048 in master. ASTERISK~28750 Change-Id: I435cea18bdd58824ed2b55259575c7ec7133842a	5 years ago
George Joseph	5f36196384	res_pjsip_outbound_registration: Fix SRV failover on timeout ... In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06	5 years ago
Joshua C. Colp	957681e08b	bridging: Add better support for adding/removing streams. ... This change adds support to bridge_softmix to allow the addition and removal of additional video source streams. When such a change occurs each participant is renegotiated as needed to reflect the update. If another video source is added then each participant gets another source. If a video source is removed then it is removed from each participant. This functionality allows you to have both your webcam and screenshare providing video if you desire, or even more streams. Mapping has been changed to use the topology index on the source channel as a unique identifier for outgoing participant streams, this will never change and provides an easy way to establish the mapping. The bridge_simple and bridge_native_rtp modules have also been updated to renegotiate when the stream topology of a party changes allowing the same behavior to occur as added to bridge_softmix. If a screen share is added then the opposite party is renegotiated. If that screen share is removed then the opposite party is renegotiated again. Some additional fixes are also included in here. Stream state is now conveyed in SDP so sendonly/recvonly/inactive streams can be requested. Removed streams now also remove previous state from themselves so consumers don't get confused. ASTERISK-28733 Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5	5 years ago
George Joseph	f93325ce52	Merge "res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough" into 17	5 years ago
Ben Ford	7a921cfe8c	RTP/ICE: Send on first valid pair. ... When handling ICE negotiations, it's possible that there can be a delay between STUN binding requests which in turn will cause a delay in ICE completion, preventing media from flowing. It should be possible to send media when there is at least one valid pair, preventing this scenario from occurring. A change was added to PJPROJECT that adds an optional callback (on_valid_pair) that will be called when the first valid pair is found during ICE negotiation. Asterisk uses this to start the DTLS handshake, allowing media to flow. It will only be called once, either on the first valid pair, or when ICE negotiation is complete. ASTERISK-28716 Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867	5 years ago
George Joseph	50704a8f4d	Merge "res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup" into 17	5 years ago
Sean Bright	5f28ecc7af	app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used ... When opening a file for writing, Asterisk silently converts filenames ending with 'wav49' to 'WAV.' We aren't taking that in to account when setting the MIXMONITOR_FILENAME variable in MixMonitor. * If the user wants to write to a wav49 file, make sure that it is reflected properly in MIXMONITOR_FILENAME. * Add a note to the documentation describing this behavior. * Add a note in main/file.c indicating that app_mixmonitor needs to be changed if the logic in build_filename was changed. ASTERISK-24798 #close Reported by: xrobau Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c	5 years ago
Torrey Searle	fb7cff4103	res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough ... When moh_passthrough is used, asterisk is only generating invites of type sendonly and sendrecv instead of taking fully into account the on hold state of the local and remote parties ASTERISK-28738 #close Change-Id: Iaaad9fbc033cb14803d433b8a4071bc337047761	5 years ago
Joshua C. Colp	5a0218a960	stasis: Use format specifier for size_t. ... Change-Id: Ic9b4afcc5398e7f46314419fc3c90433d818e35c	5 years ago
Kevin Harwell	08d8035599	res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup ... There was a race condition between client initiated DTLS setup, and handling of server side ice completion that caused the underlying SSL object to get cleared during DTLS initialization. If this happened Asterisk would be left in a partial DTLS setup state. RTP packets were sent and received, but were not being encrypted and decrypted. This resulted in no audio, or static. Specifically, this occurred when '__rtp_recvfrom' was processing the handshake sequence from the client to the server, and then 'ast_rtp_on_ice_complete' gets called from another thread and clears the SSL object when calling the 'dtls_perform_setup' function. The timing had to be just right in the sense that from the external SSL library perspective SSL initialization completed (rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks it is not finished, thus not completing 'dtls_srtp_setup'. This patch removes calls to 'dtls_perform_setup', which clears the SSL object, in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear the underlying SSL object. If an ice candidate changes a full protocol level renegotiation occurs. Also, in the case of bundled ICE candidates are reused when a stream is added. So no real reason to have to clear, and reset in this instance. Also, this patch adds a bit of extra logging to aid in diagnosis of any future problems. ASTERISK-28742 #close Change-Id: I34c9e6bad5a39b087164646e2836e3e48fe6892f	5 years ago
Joshua Colp	f424562a41	Merge "res_pjsip_session: Fix off-nominal session refreshes." into 17	5 years ago
Joshua Colp	b86ef296b3	Merge "res_musiconhold: Avoid spurious warning when 'format' is the empty string" into 17	5 years ago
George Joseph	8914a77f35	Merge "doc: Fix CHANGES entries to have .txt suffix and update READMEs" into 17	5 years ago
Sean Bright	d1a96fa9ba	res_musiconhold: Avoid spurious warning when 'format' is the empty string ... The change to res_config_odbc that allowed empty strings to be returned to realtime consumers¹ causes a warning to be emitted when loading MoH classes. So we need to treat an empty 'format' as if it was not specified to avoid the warning. ASTERISK-28735 #close Reported by: Ross Beer [1] https://gerrit.asterisk.org/c/asterisk/+/13722 Change-Id: I9a271d721e1a0973e80ebe7d75b46a0d8fa0e5a5	5 years ago
Sean Bright	4b897f6fe4	func_odbc: Prevent snprintf() truncation warning ... For reasons that are not clear to me - this only appears for me when _not_ building in dev-mode. Change-Id: Ib45c54daaea8e0d571cb470cab1daaae2edba968	5 years ago
Joshua C. Colp	3a8da5ece7	res_pjsip_session: Fix off-nominal session refreshes. ... Given a scenario where session refreshes occur close to each other while another is finishing it was possible for the session refreshes to occur out of order. It was also possible for session refreshes to be delayed for quite some time if a session refresh did not result in a topology change. For the out of order session refreshes the first session refresh would be queued due to a transaction in progress. This transaction would then finish. When finished a separate task to process the delayed requests queue would be queued for handling. A second refresh would be requested internally before this delayed request queued task was processed. As no transaction was in progress this session refresh would be immediately handled before the queued session refresh. The code will now check if any delayed requests exist before allowing a session refresh to immediately occur. If any exist then the session refresh is queued. For the delayed session refreshes if a session refresh did not result in a topology change the attempt would be immediately stopped and no other delayed requests would be processed. The code will now go through the entire delayed requests queue until a delayed request results in a request actually being sent. ASTERISK-28730 Change-Id: Ied640280133871f77d3f332be62265e754605088	5 years ago
George Joseph	4d6346dc26	doc: Fix CHANGES entries to have .txt suffix and update READMEs ... Although the wiki page for the new CHANGES and UPGRADE scheme states that the files must have the ".txt" suffix, the READMEs didn't. Change-Id: I490306aa2cc24d6f014738e9ebbc78592efe0f05 (cherry picked from commit 7416703f04)	5 years ago
Joshua Colp	47a5e33280	Merge "pjproject_bundled: Allow brackets in via parameters" into 17	5 years ago
Joshua Colp	78732ff08e	Merge "install_prereq: Install aptitude non-interactively" into 17	5 years ago
Joshua Colp	5379c065ae	Merge "chan_sip: Return 503 if we're out of RTP ports" into 17	5 years ago
Sean Bright	eb33b0fb0b	pjproject_bundled: Allow brackets in via parameters ... ASTERISK-26955 #close Reported by: Peter Sokolov Change-Id: Ib2803640905a77b65d0cee2d0ed2c7b310d470ac	5 years ago
Friendly Automation	2f992bf800	Merge "res_rtp_asterisk: Don't produce transport-cc if no packets." into 17	5 years ago
Friendly Automation	3ec3d39676	Merge "chan_sip: Clarify in sample docs how directmediapermit/-acl should be used" into 17	5 years ago
Friendly Automation	d9e625519a	Merge "res_config_odbc: Preserve empty strings returned by the database" into 17	5 years ago
Friendly Automation	259edcf5c7	Merge "res_stasis_playback: Prevent media_index from going out of bounds" into 17	5 years ago
Sylvain Afchain	cf87d63775	install_prereq: Install aptitude non-interactively ... Currently aptitude is installed using interactive mode. This patch changes this to use the non-interactive mode as it can block automatic dependencies installation, ex: CI, Docker build. ASTERISK-28726 #close Change-Id: I271ee00d230513a6f044810351a32d83b2181133 (cherry picked from commit 0c02d0a450)	5 years ago
Walter Doekes	a5ec14970d	chan_sip: Return 503 if we're out of RTP ports ... If you're for some reason out of RTP ports, chan_sip would previously responde to an INVITE with a 403, which will fail the call. Now, it returns a 503, allowing the device/proxy to retry the call on a different machine. ASTERISK-28718 Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90	5 years ago
Walter Doekes	d19dec53b5	chan_sip: Clarify in sample docs how directmediapermit/-acl should be used ... It said "restrict [...] which peers should be able to pass [audio] to each other". However, these settings are not global (for which you would expect signaling IPs to be checked). These settings are available per peer only, and the IPs being checked, are the RTP IPs. Change-Id: I2a6c6cd7c2f5f30d1df4844e3e0308a077021660	5 years ago
Joshua C. Colp	325e3f8111	res_rtp_asterisk: Don't produce transport-cc if no packets. ... The code assumed that when the transport-cc feedback function was called at least one packet will have been received. In practice this isn't always true, so now we just reschedule the sending and do nothing. Change-Id: Iabe7b358704da446fc3b0596b847bff8b8a0da6a	5 years ago
George Joseph	54cd865946	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	5 years ago
Joshua Colp	3efeb147bb	Merge "res_pjsip_messaging: Allow Content-Type to be overridden" into 17	5 years ago
Friendly Automation	e7e65588e4	Merge "res_stasis: trigger cleanup after update" into 17	5 years ago
George Joseph	2c72e26478	Merge "res_pjsip_pubsub: Increment persistence data ref when recreating." into 17	5 years ago
Friendly Automation	82eb8e7097	Merge "stasis/app: don't lock an app before a call to send" into 17	5 years ago
Sean Bright	646789106f	res_config_odbc: Preserve empty strings returned by the database ... When res_config_odbc (and perhaps other realtime backends) reads a SQL NULL from the database, it coalesces the value to the empty string which prevents it from being returned to the realtime core. However, if it instead reads the empty string from the database, it needs a way to encode that fact without having the value omitted entirely. It does this by changing the value to a string with a single space. The realtime code in main/config.c recognizes this special case and _turns the string back into the empty string_ before passing it to realtime API consumers. For all of this to work, we need to ensure that we actually pass the single-space-string back to the realtime core, which is currently failing because we are trimming the value before checking its content. So instead we now special case the single-space-string case so that empty values are returned properly. ASTERISK-28719 #close Reported by: EDV O-TON Change-Id: I673ed8c31ad037aa224e80c78c7a1dc4e4a4e3de	5 years ago
Sean Bright	8b8c1dd07f	res_stasis_playback: Prevent media_index from going out of bounds ... Incrementing stasis_app_playback.media_index directly in our playback loop means that when we reach the end of our playlist the index into the vector will be outside of the bounds of the vector. Instead use a temporary variable and only assign when we're sure that we are in bounds. ASTERISK-28713 #close Reported by: Sébastien Duthil Change-Id: Ib53f7f156097e0607eb5871d9d78d246ed274928	5 years ago
Friendly Automation	e86c32b878	Merge "res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly" into 17	5 years ago
Joshua C. Colp	35c9332edf	res_pjsip_pubsub: Increment persistence data ref when recreating. ... Each subscription needs to have a reference to the persisted data for it, as well as the main JSON contained within the tree. When recreating a subscription this did not occur and they both shared the same reference. ASTERISK-28714 Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a	5 years ago
Sean Bright	ef5702cef1	res_pjsip_messaging: Allow Content-Type to be overridden ... ASTERISK-26082 #close Reported by: Alex Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb	5 years ago
Friendly Automation	5192a045f6	Merge "chan_sip: Always process updated SDP on media source change" into 17	5 years ago