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Author	SHA1	Message	Date
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
Kevin Harwell	737bd8365e	stasis/app: don't lock an app before a call to send ... Calling 'app_send' eventually calls the app's message handler. It's possible for a handler to obtain a lock on another object, and then need/want to lock the app object. If the caller of 'app_send' locks the app object prior to calling then there's a potential for a deadlock, if another thread calls 'app_send' without locking. This patch makes it so 'app_send' is not called with the app object locked in the section of code doing such. ASTERISK-28423 #close Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27	5 years ago
Kevin Harwell	7031c3b7bd	res_stasis: trigger cleanup after update ... The cleanup code in stasis shuts down applications if they are in a deactivated state, and no longer have explicit subscriptions. When registering an app the cleanup code was running before calling 'update'. When it should be executed after 'update' since a call to register may re-activate the app. We don't want it to shutdown before the 'update' otherwise the app won't be re-activated, or registered. This patch makes it so the cleanup code is executed post 'update'. ASTERISK-28679 #close Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b	5 years ago
Sean Bright	69cf67d8c6	res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly ... We need to wait for the message sending callback to finish to know if we succeeded or failed. ASTERISK-25421 #close Reported by: Dmitriy Serov Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059	5 years ago
Walter Doekes	be6eec7b25	chan_sip: Always process updated SDP on media source change ... Fixes no-audio issues when the media source is changed and strictrtp is enabled (default). If the peer media source changes, the SDP session version also changes. If it is lower than the one we had stored, chan_sip would ignore it. This changeset keeps track of the remote media origin identifier, comparing that as well. If it changes, the session version needn't be higher for us to accept the SDP. Common scenario where this would've caused problems: a separate media gateway that informs the caller about premium rates before handing off the call to the final destination. (An alternative fix would be to set ignoresdpversion=yes on the peer.) ASTERISK-28686 Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee	5 years ago
Sean Bright	ace97a3786	chan_pjsip: Ignore RTP that we haven't negotiated ... If chan_pjsip receives an RTP packet whose payload differs from the channel's native format, and asymmetric_rtp_codec is disabled (the default), Asterisk will switch the channel's native format to match that of the incoming packet without regard to the negotiated payloads. We now check that the received frame is in a format we have negotiated before switching payloads which results in these packets being dropped instead of causing the session to terminate. ASTERISK-28139 #close Reported by: Paul Brooks Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3	5 years ago
Asterisk Development Team	a0f44b1052	Update CHANGES and UPGRADE.txt for 17.2.0	5 years ago
Friendly Automation	6747dd31b4	Merge "http: Add ability to disable /httpstatus URI" into 17	5 years ago
George Joseph	f34a4d5a77	cdr.c: Set event time on party b when leaving a parking bridge ... When Alice calls Bob and Bob does a blind transfer to Charlie, Bob's bridge leave event generates a finalize on both the party_a and party_b CDRs but while the party_a CDR has the correct end time set from the event time, party_b's leg did not. This caused that CDR's end time to be equal to the answered time and resulted in a billsec of 0. * We now pass the bridge leave message event time to cdr_object_party_b_left_bridge_cb() and set it on that CDR before calling cdr_object_finalize() on it. NOTE: This issue affected transfers using chan_sip most of the time but also occasionally affected chan_pjsip probably due to message timing. ASTERISK-28677 Reported by: Maciej Michno Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca	5 years ago
Sean Bright	1e79bf44fc	http: Add ability to disable /httpstatus URI ... Add a new configuration option 'enable_status' which allows the /httpstatus URI handler to be administratively disabled. We also no longer unconditionally register the /static and /httpstatus URI handlers, but instead do it based upon configuration. Behavior change: If enable_static was turned off, the URI handler was still installed but returned a 403 when it was accessed. Because we now register/unregister the URI handlers as appropriate, if the /static URI is disabled we will return a 404 instead. Additionally: * Change 'enablestatic' to 'enable_static' but keep the former for backwards compatibility. * Improve some internal variable names ASTERISK-28710 #close Change-Id: I647510f796473793b1d3ce1beb32659813be69e1	5 years ago
Friendly Automation	f7e0b6baf8	Merge "func_odbc.conf.sample: Add example lookup" into 17	5 years ago
Friendly Automation	f556efe0b9	Merge "res_statsd: Document that res_statsd does nothing on its own" into 17	5 years ago
Friendly Automation	c8dc9365a4	Merge "translate.c: Fix silk 24kHz truncation in 'core show translation'" into 17	5 years ago
Joshua Colp	73cd259895	Merge "chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout"" into 17	5 years ago
Friendly Automation	d6a4560016	Merge "chan_sip.c: Stop handling continuation lines after reading headers" into 17	5 years ago
Andrew Siplas	1c5116f842	chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout" ... The no-entry timeout set to 999999 == 16⅔ minutes, change to INT_MAX to match behavior of "no timeout" defined in comment. ASTERISK-28702 #close Change-Id: I4ea015986e061374385dba247b272f7aac60bf11	5 years ago
Sean Bright	687890f734	translate.c: Fix silk 24kHz truncation in 'core show translation' ... SILK @ 24kHz is not shown in the 'core show translation' output because of an off-by-one-error. Discovered while looking into ASTERISK~19871. ASTERISK-28706 Reported by: Sean Bright Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510	5 years ago
Sean Bright	d33cd39260	func_odbc.conf.sample: Add example lookup ... Change-Id: Ia05aab1f579597963d2ea23920d2210cfcb97c84	5 years ago
Sean Bright	deed0f5706	res_statsd: Document that res_statsd does nothing on its own ... ASTERISK-24484 #close Reported by: Dan Jenkins Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec	5 years ago
Friendly Automation	503cac85ee	Merge "queue_log: Add alembic script for generate db table for queue_log" into 17	5 years ago