31695 Commits (02c7a061ea312efd6401be9ffa0bcffc03f2f0e2)

 

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Author	SHA1	Message	Date
Corey Farrell	02c7a061ea	res_pjsip_caller_id: Use static pj_str_t for fromto header names. ... PJSIP assumes that these header names are not allocated, does not clone the name strings when reusing headers. Block unload of res_pjsip_caller_id until shutdown to ensure static memory stays valid. It was previously unsafe to unload while any sessions are active. Change-Id: I190854dea943d6e441cf03733f8a0da661aea27f	8 years ago
George Joseph	fc82312aab	Merge "pbx_config: Only the first [globals] section is seen."	8 years ago
George Joseph	eb5b83b8ea	AST-2018-010: Fix length of buffer needed for SRV and NAPTR results ... When dn_expand was being called on SRV and NAPTR results, the return value was being used to calculate the size of the buffer needed to store the host names. Since dn_expand returns the length of the COMPRESSED name the buffer could be too short to hold the EXPANDED name. The expanded name is NULL terminated so using strlen() is the correct way to determine the length actually needed for the buffer. ASTERISK-28127 Reported by: Jan Hoffmann patches: patch.diff submitted by janhoffmann (license 6986) Change-Id: I4d35d6c431c6c6836cb61d37b1378cc47f0b414d	8 years ago
Joshua Colp	9e151413db	Merge "core: Ensure that el_end is always run when needed."	8 years ago
Joshua Colp	dbfc5ae020	Merge "taskprocessor: Do not use separate allocation for stats or name."	8 years ago
George Joseph	39cfc56dd8	Merge "jansson-bundled: Patch for off-nominal crash."	8 years ago
Corey Farrell	944d90a7ea	taskprocessor: Do not use separate allocation for stats or name. ... Merge storage for the stats object and name string into the main allocation for struct ast_taskprocessor. Change-Id: I74fe9a7f357f0e6d63152f163cf5eef6428218e1	8 years ago
Joshua Colp	6f3275e54c	Merge "res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue"	8 years ago
Corey Farrell	194e40122a	core: Ensure that el_end is always run when needed. ... * Ignore console=yes configuration option in remote console processes. * Use new flag to tell consolethread to run el_end and exit when needed. ASTERISK-28158 Change-Id: I9e23b31d4211417ddc88c6bbfd83ea4c9f3e5438	8 years ago
Corey Farrell	d9add7e086	jansson-bundled: Patch for off-nominal crash. ... pack_string crashed on non-NULL strings returned when s->has_error was true if the string was the result of 's' format without '#', '%' or '+'. Change-Id: Ic125df691d81ba2cbc413e37bdae657b304d20d0	8 years ago
Corey Farrell	8e34cb302e	pbx_config: Only the first [globals] section is seen. ... If multiple [globals] sections are used (for example via separate included files), only the first one is processed. This can result in lost global variables when using a modular extensions.conf. ASTERISK-28146 #close Change-Id: Iaac810c0a7c4d9b1bf8989fcc041cdb910ef08a0	8 years ago
Chris-Savinovich	a3fc97aa13	res_pjsip: Send a 503 response when overload state if reliable transport. ... When Asterisk's taskprocessors get overloaded we need to reduce the work load. res_pjsip currently ignores new SIP requests and relies on SIP retransmissions in the hope that the overload condition will clear soon enough to handle the retransmitted SIP request. This change adds the following code after ast_taskprocessor_alert_get() has returned TRUE: 1- identifies transport type. If non-udp then send a 503 response 2- if transport type is udp/udp6 then ignore, as before. Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836	8 years ago
Joshua Colp	d3bc9e6784	Merge "stasis: Clarify lifetime of topics."	8 years ago
Kevin Harwell	fdca9cb64f	res_pjsip: formatting error in documentation ... The use of a '|' in the "global/debug" synopsis documentation caused the generated html table on the wiki to add an extra column that included the text after the pipe. This patch replaces the pipe with a comma. ASTERISK-28150 Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719	8 years ago
Alexei Gradinari	5f3f707793	res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue ... The current round-robin method does not take the current taskprocessor load into consideration when distributing requests. Using the least-size method the request goes to the taskprocessor that is servicing the least number of active tasks at the current time. Longer running tasks with the round-robin method can delay processing tasks. * Change the algorithm from round-robin to least-size for picking the PJSIP taskprocessor from the default serializer pool. Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd	8 years ago
George Joseph	65cbf60efc	Merge "chan_sip: Attempt ast_do_pickup in handle_invite_replaces"	8 years ago
Joshua Colp	bf579222c4	stasis: Clarify lifetime of topics. ... As mentioned in the comment I've added in the code there is no ability to unsubscribe all subscribers from a topic and explicitly destroy it. This is not currently a problem as we have two types of topics: Long lived topics which exist for the lifetime of the system. Ephemeral topics which feed a long lived topic. In the case of the ephemeral topics there is no subscriber which does not have its lifetime managed by the same entity that has created the topic. This ensures that when the topic is being unreferenced the subscribers are also unsubscribed and destroyed, allowing the topic to ultimately be destroyed as well. Change-Id: Ic5e244da7b16b1895ba1fc5ece481ebba5809c9a	8 years ago
Jasper Hafkenscheid	2cf5079205	chan_sip: Attempt ast_do_pickup in handle_invite_replaces ... When a call pickup is performed using and invite with replaces header the ast_do_pickup method is attempted and a PICKUP stasis message is sent. ASTERISK-28081 #close Reported-by: Luit van Drongelen Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a	8 years ago
Pascal Cadotte Michaud	ebff81e3a0	contrib/sip_to_pjsip: add a --quiet option to avoid prints ... Using the --quiet or -q option in conjonction with /dev/stdout as the output file allow the output to be used as a valid configuration. Given a script that generates a valid sip.conf I can pipe the output of that script into `sip_to_pjsip.py -q /dev/stdin /dev/stdout`. This allow me to use that piped command in my pjsip.conf using the `exec` command. ASTERISK-28136 Change-Id: I7b0e2e90e2549f3f8e01dc96701f111b5874c88d	8 years ago
George Joseph	7ece4af59b	Merge "res_pjsip: Add XML documentation for "use_callerid_contact""	8 years ago
George Joseph	5ea16f0cb5	Merge "alembic: Fix use_callerid_contact option add script."	8 years ago
George Joseph	26810197c7	Merge "pjsip: new endpoint's options to control Connected Line updates"	8 years ago
George Joseph	bcd07a86ba	Merge "contrib/sip_to_pjsip: handle setvar in conversion"	8 years ago
George Joseph	f1f4450bdf	Merge "chan_sip deprecation."	8 years ago
Joshua Colp	0c9e217c81	res_pjsip: Add XML documentation for "use_callerid_contact" ... ASTERISK-28087 Change-Id: I69d48813ec514f5ef06c6de994cba52630e0a3b4	8 years ago
Richard Mudgett	c7528f16e6	alembic: Fix use_callerid_contact option add script. ... ASTERISK-28087 Change-Id: I046d018015427d0916fab571b5a4f5367476f729	8 years ago
Alexei Gradinari	eee935983b	pjsip: new endpoint's options to control Connected Line updates ... This patch adds new options 'trust_connected_line' and 'send_connected_line' to the endpoint. The option 'trust_connected_line' is to control if connected line updates are accepted from this endpoint. The option 'send_connected_line' is to control if connected line updates can be sent to this endpoint. The default value is 'yes' for both options. Change-Id: I16af967815efd904597ec2f033337e4333d097cd	8 years ago
Pascal Cadotte Michaud	b0155f7e58	contrib/sip_to_pjsip: handle setvar in conversion ... Given a sip.conf with the following content: setvar FOO=1 setvar BAR=42 I want my generated pjsip.conf to containt the following set_vars set_var FOO=1 set_var BAR=42 in the matching endpoint section. Change-Id: I6c822401fda4133c3b44bf31e655b4eb939d4d26	8 years ago
George Joseph	584e08b81b	Merge "res_pjsip_notify: improve realtime performance on CLI completion on the endpoint"	8 years ago
Alexei Gradinari	e407b8af21	res_pjsip_notify: improve realtime performance on CLI completion on the endpoint ... The module 'res_pjsip_notify' inefficiently makes a lot of DB requests on CLI completion on the endpoint. For example if there are 10k endpoints the module makes 10k requests of these 10k records. Even if a part of the endpoint entered the module makes the same 10k requests and then filtered them by itself. This patch gathers endpoints container by prefix and adds all gathered endpoints to completion at once. ASTERISK-28137 #close Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b	8 years ago
Torrey Searle	cac4ccef25	res_pjsip_session: add new flag use_callerid_contact ... Add a new global flag to res_pjsip to allow the callerid to be used as the username in the contact header. This allows chan_pjsip to have the same behavour as chan_sip ASTERISK-28087 #close Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95	8 years ago
Corey Farrell	90a11c4ae7	chan_sip deprecation. ... This officially deprecates chan_sip in Asterisk 17+. A warning is printed upon startup or module load to tell users that they should consider migrating. chan_sip is still built by default but the default modules.conf skips loading it at startup. Very important to note we are not scheduling a time where chan_sip will be removed. The goal of this change is to accurately inform end users of the current state of chan_sip and encourage movement to the fully supported chan_pjsip. Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93	8 years ago
Corey Farrell	e81d33e78f	UPDATE.txt: Fix formatting to match previous files. ... Add 'Section:' headings and use '-' for bullet points. Change-Id: I7e2be35601ac8fea53b90d926da564512b6716e4	8 years ago
Joshua Colp	785669e658	Merge "res_parking: Stop setting the deprecated PARKINGSLOT channel variable."	8 years ago
Joshua Colp	e80f2012e6	Merge "app_dial/queue/followme: 'I' options to block initial updates in both directions"	8 years ago
Joshua Colp	e7b22ee133	Merge "bridge_softmix: Add SDP "label" attribute to streams"	8 years ago
George Joseph	07c950b2dd	Merge "say: Remove legacy language deprecation logic"	8 years ago
George Joseph	780c260219	Merge "logger.c: Fix default console logging when no logger.conf available."	8 years ago
Joshua Colp	bf5bb7831f	Merge "modules.conf.sample: Update preload usage documentation."	8 years ago
Sean Bright	79c2b4fddd	res_parking: Stop setting the deprecated PARKINGSLOT channel variable. ... Change-Id: Ia155ce2a53d61556aa4685524d1b48cfacfa3a8b	8 years ago
Joshua Colp	12643d3d7b	Merge "func_callerid: Remove deprecated CALLERPRES() function."	8 years ago
Joshua Colp	dbfb75e02d	Merge "res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability"	8 years ago
Richard Mudgett	1b397ebd00	logger.c: Fix default console logging when no logger.conf available. ... Default logging was not setup correctly when there was no logger.conf. This resulted in many expected log messages not actually getting out to the console. Change-Id: I542e61c03b2f630ff5327f9de5641d776c6fa70c	8 years ago
Alexei Gradinari	4a567cee3a	app_dial/queue/followme: 'I' options to block initial updates in both directions ... The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates from the called parties to the caller. This patch also blocks updates in the other direction before call is answered. ASTERISK-27980 Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01	8 years ago
Richard Mudgett	96d5e444f0	modules.conf.sample: Update preload usage documentation. ... Change-Id: Id449d4435c38148b56ac4cfd61ae4d90ac66bb90	8 years ago
George Joseph	8d1c6bb6e6	bridge_softmix: Add SDP "label" attribute to streams ... Adding the "label" attribute used for participant info correlation was previously done in app_confbridge but it wasn't working correctly because it didn't have knowledge about which video streams belonged to which channel. Only bridge_softmix has that data so now it's set when the bridge topology is changed. ASTERISK-28107 Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499	8 years ago
George Joseph	a99d48d3f3	Merge "astobj2: Eliminate legacy container allocation macros."	8 years ago
Sean Bright	056ca07449	func_callerid: Remove deprecated CALLERPRES() function. ... Change-Id: Ia1b2b386505b3102136dab02c45eaaf09f0f89c5	8 years ago
Nick French	37b2e68628	res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability ... This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58	8 years ago
George Joseph	51b5f0f193	Merge "res_xmpp: Remove deprecated JabberStatus application."	8 years ago