32677 Commits (00a52b4752497a6e9e942d70a5e07566a1d6fe9c)

 

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Author	SHA1	Message	Date
Joshua Colp	0c845063b3	Merge "res_pjsip_messaging: Allow Content-Type to be overridden"	6 years ago
Walter Doekes	43620cbf6c	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	6 years ago
Friendly Automation	d4b9529a58	Merge "res_stasis: trigger cleanup after update"	6 years ago
Friendly Automation	87ab83d03f	Merge "stasis/app: don't lock an app before a call to send"	6 years ago
Friendly Automation	499f41051e	Merge "res_pjsip_pubsub: Increment persistence data ref when recreating."	6 years ago
Sean Bright	eb9252ea27	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	6 years ago
Sean Bright	31dc904380	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	6 years ago
Friendly Automation	e2119e8968	Merge "res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly"	6 years ago
Joshua C. Colp	a1f0c833ab	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	6 years ago
Sean Bright	03d24ca4c1	res_pjsip_messaging: Allow Content-Type to be overridden ... ASTERISK-26082 #close Reported by: Alex Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb	6 years ago
Walter Doekes	113d05e504	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	6 years ago
Friendly Automation	f29ddd8925	Merge "chan_sip: Always process updated SDP on media source change"	6 years ago
Kevin Harwell	cce2b0da95	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	6 years ago
Kevin Harwell	4206830a52	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	6 years ago
Sean Bright	b1ca2c5d71	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	6 years ago
Walter Doekes	711a3fed56	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	6 years ago
Sean Bright	313189aae2	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	6 years ago
George Joseph	9688381f05	Merge "http: Add ability to disable /httpstatus URI"	6 years ago
Joshua Colp	b88791d88c	Merge "cdr.c: Set event time on party b when leaving a parking bridge"	6 years ago
Friendly Automation	95c6fbeae0	Merge "app_voicemail: Remove MessageExists and MESSAGE_EXISTS()"	6 years ago
George Joseph	6818c3d1d2	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	6 years ago
Sean Bright	0dce6f746b	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	6 years ago
Joshua Colp	b073b4318a	Merge "func_odbc.conf.sample: Add example lookup"	6 years ago
Friendly Automation	9c1462f0e4	Merge "res_statsd: Document that res_statsd does nothing on its own"	6 years ago
Joshua Colp	d20c7ed807	Merge "translate.c: Fix silk 24kHz truncation in 'core show translation'"	6 years ago
Joshua Colp	093f349daf	Merge "chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout""	6 years ago
Friendly Automation	f1e06a4528	Merge "chan_sip.c: Stop handling continuation lines after reading headers"	6 years ago
Andrew Siplas	5bd7281442	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	6 years ago
Sean Bright	c376e9f8a8	res_statsd: Document that res_statsd does nothing on its own ... ASTERISK-24484 #close Reported by: Dan Jenkins Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec	6 years ago
Sean Bright	dfad69ce7c	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	6 years ago
Sean Bright	262221f4d9	func_odbc.conf.sample: Add example lookup ... Change-Id: Ia05aab1f579597963d2ea23920d2210cfcb97c84	6 years ago
Joshua Colp	dbb3a19c35	Merge "queue_log: Add alembic script for generate db table for queue_log"	6 years ago
Joshua Colp	64debbd13f	Merge "app_voicemail, say: Fix various leading whitespace problems"	6 years ago
Joshua Colp	058e9f735e	Merge "app_voicemail: Prevent crash when saving message with realtime voicemail"	6 years ago
Joshua Colp	8208f5c73d	Merge "pbx.c: Include filesystem cache in free memory calculation"	6 years ago
George Joseph	0380288f7c	Merge "res_realtime: Fix 'realtime update2' argument handling"	6 years ago
Joshua Colp	2d17e25015	Merge "app_voicemail: Set globals to default values when voicemail.conf missing"	6 years ago
Sean Bright	f09cf4da44	app_voicemail: Remove MessageExists and MESSAGE_EXISTS() ... * The MailboxExists dialplan application was deprecated on 2006-09-26 in Asterisk 1.6.0 (commit ec83b11183) * The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in Asterisk 11.0.0 (commit fd64bb66f9) Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f	6 years ago
Sean Bright	5cbf47714a	app_voicemail, say: Fix various leading whitespace problems ... In af90afd90c, Japanese language support was added to app_voicemail and main/say.c, but the leading whitespace is not consistent with Asterisk coding guidelines. This patch fixes that. Whitespace only, no functional change. ASTERISK~23324 Reported by: Kevin McCoy Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87	6 years ago
Sean Bright	50d02d6194	pbx.c: Include filesystem cache in free memory calculation ... ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	6 years ago
Sean Bright	f309b86e36	chan_sip.c: Stop handling continuation lines after reading headers ... lws2sws() does not stop trying to handle header continuation lines even after all headers have been found. This is problematic if the first character of a SIP message body is a space or tab character, so we update to recognize the end of the message header. ASTERISK-28693 #close Reported by: Frank Matano Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df	6 years ago
Sean Bright	ba8ccb9132	app_voicemail: Prevent crash when saving message with realtime voicemail ... ast_store_realtime() is not NULL tolerant, so we need to initialize the field values we pass to it to the empty string to avoid a crash. ASTERISK-23739 #close Reported by: Stas Kobzar Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c	6 years ago
Friendly Automation	4255277ffd	Merge "feat: AudioSocket channel, application, and ARI support."	6 years ago
Joshua Colp	d5fce4bc34	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY"	6 years ago
Friendly Automation	c665878e92	Merge "app_queue: Deprecate the QueueMemberPause.Reason field"	6 years ago
Sean Bright	9be89d9913	app_voicemail: Set globals to default values when voicemail.conf missing ... If voicemail.conf exists but is empty, the config parsing process will default a number of global variables to non-zero values. On the other hand, if voicemail.conf is missing (arguably semantically equivalent to an empty file), this process is skipped and the globals are defaulted to 0. Set the globals to the same values they would be set to if a configuration were present. This allows voicemail configuration to be done completely by Realtime without the need to create an empty voicemail.conf file. ASTERISK-27622 #close Reported by: Jim Van Meggelen Change-Id: Id907d280f310f12e542ca527e6a025432b9fb409	6 years ago
Friendly Automation	3098adc816	Merge "res_pjsip_endpoint_identifier_ip: Document support for hostnames"	6 years ago
Joshua Colp	be284af0d3	Merge "func_curl: Add 'followlocation' option to CURLOPT()"	6 years ago
Sean Bright	094e87b0dc	res_realtime: Fix 'realtime update2' argument handling ... The change in 9b99ef50b5 updated the syntax of the 'realtime update2' CLI command but did not correctly update the calls to ast_update2_realtime(). The issue this addresses was originally opened because we aren't allowing a SQL NULL to be set as part of the update, but this is a limitation of the Realtime API and is not a bug. Additionally, this patch: * Corrects the example in the command documentation to reflect 'update2' instead of 'update.' * Fixes the leading spacing of the command documentation. * Checks that the required 'NULL' literal argument is present where we expect it to be. ASTERISK-21794 #close Reported by: Cédric Bassaget Change-Id: Idda63a5dc50d5f9bcb34c27ea3238d90f733b2cd	6 years ago
Friendly Automation	3f663a543d	Merge "netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr."	6 years ago