4405 Commits (ba068d3576ffb384392dff131b58396bd32d858c)

Author	SHA1	Message	Date
George Joseph	ba068d3576	Merge "res/res_pjsip_nat: Fix logic for REINVITES" into 13	7 years ago
Torrey Searle	da4879443b	res/res_pjsip_nat: Fix logic for REINVITES ... The presence of Record-Route in re-invites is optional, thus it is important to make sure the dialog doesn't have a routset before rewriting the contact header. ASTERISK-28129 #close Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc	7 years ago
Joshua Colp	6aea312a55	Merge "res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue" into 13	7 years ago
Chris-Savinovich	bc2420ee0e	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	7 years ago
Kevin Harwell	214d0d118a	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	7 years ago
Alexei Gradinari	158214c1a0	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	7 years ago
George Joseph	5605928db0	Merge "res_pjsip: Add XML documentation for "use_callerid_contact"" into 13	7 years ago
Joshua Colp	9946bcc557	res_pjsip: Add XML documentation for "use_callerid_contact" ... ASTERISK-28087 Change-Id: I69d48813ec514f5ef06c6de994cba52630e0a3b4	7 years ago
Alexei Gradinari	bfe3821800	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	7 years ago
George Joseph	0b695d1e38	Merge "res_pjsip_notify: improve realtime performance on CLI completion on the endpoint" into 13	7 years ago
Alexei Gradinari	82f0a86e39	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	7 years ago
Torrey Searle	bbbec2e95e	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	7 years ago
George Joseph	1c33cf5cf1	Merge "astobj2: Eliminate usage of legacy container allocation macros." into 13	7 years ago
Corey Farrell	54a1fbe428	astobj2: Eliminate usage of legacy container allocation macros. ... These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are still available for use but only in modules. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
Richard Mudgett	d8b9e3413c	Fix 'statement' typo throughout code. ... Most were in comments. A couple were in warning messages. Pointed out by Jonathan H on the Asterisk users mailing list. Change-Id: I6286939dff5d0a27a2758140570106f1cb351855	7 years ago
George Joseph	6296b5c7be	Merge "res_pjsip: set callerid_tag to empty string" into 13	7 years ago
Alexei Gradinari	6b1a981ee1	res_pjsip: set callerid_tag to empty string ... This patch sets the callerid_tag to empty string by default. If the callerid_tag is set to NULL then the tag does not become part of a connected line update. For example: Alice's tag is "Alice". Bob's tag is empty. Charlie's tag is "Charlie". Alice calls Bob and then does attended transfer to Charlie. When Alice hangs up the CONNECTEDLINE(tag) is "Alice" on the interception routine on the Charlie's channel, but should be empty. Ths patch also fix memory leaks if there are more then one options "callerid", "callerid_tag", "voicemail_extension" and "contact_user" in the pjsip.conf endpoint definition. Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4	7 years ago
Richard Mudgett	ccdb53a1ca	res_statsd.c: Fix returned reload status. ... The return status when there was no change in statsd.conf was incorrect. This resulted in the wrong status message on the CLI when reloading the module. * Fixed cleanup on initial load if initializing statsd failed. Change-Id: Id24fae75f1a7ff584a444a5680e867d989792481	7 years ago
Richard Mudgett	d55ded340f	res_smdi.c: Fix module ref counting and inverted test. ... I think this module is so screwed up that it doesn't work anymore. Even with these attempts to fix things it still won't gracefully shut down. The module refs will not go to zero to allow unloading the module. * Fix module ref counting dealing with the SMDI interface object. There were several off-nominal paths that unbalanced the module ref count. Also the destructor freed the ao2 object itself which is bad. Made the smdi_read thread not hold its own ref to the SMDI interface object so when all refs go away the destructor will stop the listener thread. * Fixed the smdi_load() return code of 1 concerning the number of listeners. The test was inverted. Change-Id: Ic288db51b58e395d6a2fc3847f77176c16988784	7 years ago
Joshua Colp	a02522c140	Merge "res_pjsip: improve realtime performance on CLI 'pjsip show contacts'" into 13	7 years ago
Joshua Colp	dc03526380	Merge "res_stasis: Fix stale data in ARI bridges" into 13	7 years ago
Kevin Harwell	fa76297e66	Merge "res_odbc: fix missing SQL error diagnostic" into 13	7 years ago
Ben Ford	aa31657e28	res_rtp_asterisk.c: Add "seqno" strictrtp option ... When networks experience disruptions, there can be large gaps of time between receiving packets. When strictrtp is enabled, this created issues where a flood of packets could come in and be seen as an attack. Another option - seqno - has been added to the strictrtp option that ignores the time interval and goes strictly by sequence number for validity. Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71	7 years ago
Alexei Gradinari	7d79805d54	res_pjsip: improve realtime performance on CLI 'pjsip show contacts' ... CLI command 'pjsip show contacts' inefficiently make a lot of DB requests. For example if there are 10k aors then asterisk requests these 10k records of aor and then does 10k requests of contact - one request per aor. Even if use 'like <pattern>' the asterisk requests all aor's and contact's records and then filters them by itself. This patch gathers contact's container by - retrieving all dynamic contacts by regex (filtered by reg_server) - retrieving all aors with permanent contacts - finally filters container by regex ASTERISK-28077 #close Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618	7 years ago
George Joseph	efadf89131	Merge "res_rtp_asterisk: Raise event when RTP port is allocated" into 13	7 years ago
Moritz Fain	27c46b8ff3	res_stasis: Fix stale data in ARI bridges ... Fixed an issue that resulted in "Allocation failed" each time an ARI request was made to start playing MOH on a bridge. In bridge_moh_create() we were attaching the after bridge callbacks to chan which is the ;1 channel of the unreal channel pair. We should have attached them to the ;2 channel which is pushed into the bridge by ast_unreal_channel_push_to_bridge(). The callbacks are called when the specific channel leaves the bridging system. Since the ;1 channel is never put into a bridge the callbacks never get called. The callbacks then never remove the moh_wrapper from the app_bridges_moh container. As a result we cannot find the channel associated with the wrapper to start MOH because it has hungup. This is the reason causing the reported issue. * Rather than using after bridge callbacks to cleanup, we now have moh_channel_thread() doing the cleanup when the channel hangs up. * Fixed moh_channel_thread() accumulating control frames on the stasis bridge MOH channel until MOH is stopped. Control frames are no longer accumulated while MOH is playing. * Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may or may not return a channel ref. As a result ast_ari_bridges_start_moh() wouldn't know it may have a channel ref to release. stasis_app_bridge_moh_channel() will now return a ref with the channel it returns. * Eliminated RAII_VAR in bridge_moh_create(). ASTERISK-26094 #close Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c	7 years ago
Joshua Colp	155ff8e174	res_rtp_asterisk: Raise event when RTP port is allocated ... This change raises a testsuite event to provide what port Asterisk has actually allocated for RTP. This ensures that testsuite tests can remove any assumption of ports and instead use the actual port in use. ASTERISK-28070 Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044	7 years ago
Alexei Gradinari	37c1f35922	res_odbc: fix missing SQL error diagnostic ... On SQL error there is not diagnostic information about this error. There is only WARNING res_odbc.c: SQL Execute error -1! The function ast_odbc_print_errors calls a SQLGetDiagField to get the number of available diagnostic records, but the SQLGetDiagField returns 0. However SQLGetDiagRec could return one diagnostic records in this case. Looking at many example of getting diagnostics error information I found out that the best way it's to use only SQLGetDiagRec while it returns SQL_SUCCESS. Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR to res_config_odbc. ASTERISK-28065 #close Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1	7 years ago
Sean Bright	3033242a7b	res_rtp_asterisk: Reset all settings on module reload ... 'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults if they are not present in the updated configuration file. Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670	7 years ago
Sean Bright	33ca3664ca	AST-2018-009: Fix crash processing websocket HTTP Upgrade requests ... The HTTP request processing in res_http_websocket allocates additional space on the stack for various headers received during an Upgrade request. An attacker could send a specially crafted request that causes this code to overflow the stack, resulting in a crash. * No longer allocate memory from the stack in a loop to parse the header values. NOTE: There is a slight API change when using the passed in strings as is. We now require the passed in strings to no longer have leading or trailing whitespace. This isn't a problem as the only callers have already done this before passing the strings to the affected function. ASTERISK-28013 #close Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a	7 years ago
George Joseph	c4f1adf78e	Merge "pjproject: Upgrade to 2.8." into 13	7 years ago
Joshua Colp	cd2deadb69	pjproject: Upgrade to 2.8. ... This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	7 years ago
Sean Bright	99b2e0c2ff	autoconf: Check for srtp_get_version_string() before using it ... Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df	7 years ago
George Joseph	926ac196af	Merge "res_srtp.c: Show linked version of libsrtp on module init" into 13	7 years ago
George Joseph	3a09d9c74c	Merge "res_pjsip: Log IPv6 addresses correctly" into 13	7 years ago
Sean Bright	44375c0616	res_srtp.c: Show linked version of libsrtp on module init ... Change-Id: Ib0a645d6985de5757cc4399ed2524b2d02c4f342	7 years ago
Sean Bright	d3c869c736	res_pjsip: Log IPv6 addresses correctly ... Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name store IPv6 addresses without enclosing brackets. This causes some log output to be confusing because it is difficult to separate the IPv6 address from a port specification. * Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6 output. * When a pj_sockaddr is not available, explicitly wrap IPv6 addresses in brackets. * When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure to also set pjsip_rx_data.pkt_info.src_addr. Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8	7 years ago
Jenkins2	d4becda8ec	Merge "res_musiconhold.c: Restart MOH if previous hold just reached end-of-file" into 13	7 years ago
Jenkins2	641e5b7e63	Merge "optional_api: Remove unused nonoptreq fields" into 13	7 years ago
Walter Doekes	d226458c5b	optional_api: Remove unused nonoptreq fields ... As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc	7 years ago
Sean Bright	931b68d3e6	res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP ... The bundled version of pjproject has a patch for Solaris compatability that changes the definition of various socket structures which we need to account for when compiling against a non-bundled version. ASTERISK-28049 #close Change-Id: Ia1ea47c433fc2d915115193ee889a752373925f0	7 years ago
Frederic LE FOLL	a5f87a3a36	res_musiconhold.c: Restart MOH if previous hold just reached end-of-file ... On MOH activation, moh_files_readframe() is called while the current stream attached to the channel is NULL and it calls ast_moh_files_next() immediately. However, it won't call ast_moh_files_next() again if sample reading fails. The failure may occur because res_musiconhold retains the last sample reading position in the channel data and MOH during the previous hold/retrieve just reached EOF. Obviously, a bit of bad luck is required here. * Restructured moh_files_readframe() to try a second time to start MOH if there was no stream setup and the saved position was at EOF. Also added comments describing what is going on for each step. ASTERISK-28029 Change-Id: I1508cf2c094f8feca22d6f76deaa9fdfa9944860	7 years ago
Sean Bright	b0ef54af71	res_pjproject: Add utility functions to convert between socket structures ... Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address needs to be rendered to a string and then parsed into the correct structure. This also involves a call to getaddrinfo(3). The same is true for the inverse operation. Instead, because we know the internal structure of both ast_sockaddr and pj_sockaddr, we can translate directly between the two without the need for an intermediate string. Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761	7 years ago
George Joseph	5d4cf16c17	Merge "res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch" into 13	7 years ago
Joshua Colp	d0da68bfc9	Merge "res_pjsip_transport_websocket: Properly set src_name for IPv6" into 13	7 years ago
Joshua Colp	6bf0943ecb	Merge "res_fax: Handle fax gateway being started more than once." into 13	7 years ago
Richard Mudgett	f2266602a1	res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch ... ASTERISK-27988 Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843	7 years ago
Joshua Colp	cfe8ef917d	Merge "Create --disable-binary-modules option." into 13	7 years ago
Joshua Colp	fa71f498ce	res_fax: Handle fax gateway being started more than once. ... The T.38 fax gateway state machine can cause the fax gateway to be started more than once on a channel depending on the responses of the remote endpoint. This would previously leak the channel name, channel unique id, and underlying fax engine state. This change instead makes it so that if the fax gateway session is already present and not reserved the fax gateway is not started again. ASTERISK-27981 Change-Id: I552d95086860cb18f2522ee40ef47b13b6da2e0e	7 years ago
Sean Bright	f3188a634d	res_pjsip_transport_websocket: Properly set src_name for IPv6 ... SIP responses over WebSockets when the client is using IPv6 have invalid Via headers according to RFC 3261. The 'received' header parameter should not be wrapped in brackets if it is an IPv6 address. When src_name is populated by the built-in PJSIP transports, the code uses pj_sockaddr_print() with 'flags' set to 0, meaning that the brackets are not rendered around IPv6 addresses. This may be related to ASTERISK~27101. See also: https://github.com/onsip/SIP.js/pull/594 ASTERISK-28020 #close Change-Id: I8ea9d289901b837512bee2ca2535e3dc14f04d77	7 years ago