asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	ead0bc63da	Merge "http.c: Give HTTP error response when received lines are too long." into 16	7 years ago
George Joseph	9fb166cf3b	stasis_cache: Prune stasis_subscription_change messages The stasis cache provides a way to reconstruct the current state of topic subscribers. Unfortunately, since every subscribe and unsubscribe is cached, the cache continues to grow unabated while asterisk is running. This patch removes subscribe messages from the cache when the corresponding unsubscribe is received. This patch also registers the cache containers with ao2 so that if AO2_DEBUG is turned on, you can list the container and get its stats from the CLI. ASTERISK-27121 Change-Id: I3d18905e477f3721815da91f30da8d3fbb2d4f56	7 years ago
George Joseph	85a7c33acf	Merge "app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done" into 16	7 years ago
George Joseph	597f612645	Merge "res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch" into 16	7 years ago
George Joseph	10460501ca	Merge "iostream.c: Fix ast_iostream_gets() needlessly returning failure." into 16	7 years ago
Rodrigo Ramírez Norambuena	8879a62c1c	app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8	7 years ago
Chris-Savinovich	cfb854e241	pbx_config.c: Fix reloading module if initially declined to load Added decline if extensions.conf file not available when loading pbx_config, and also made sure everything gets properly unregistered and/or destroyed on unload. Change-Id: Ib00665106043b1be5148ffa7a477396038915854	7 years ago
Joshua Colp	ed7cef7d06	Merge "make config: os-release output error." into 16	7 years ago
Richard Mudgett	4fcdcfaa37	http.c: Give HTTP error response when received lines are too long. Added a check when we receive a HTTP request line or header line that is too long. We now return an error response to the sender because we are not able to process the request. Change-Id: I6df2705435fd7dde4d5d3bdf7acec859cfb7c12d	7 years ago
Richard Mudgett	f6a165208b	iostream.c: Fix ast_iostream_gets() needlessly returning failure. Providing a buffer larger than the internal buffer of ast_iostream_gets() fails to get lines longer than the internal buffer. * Made ast_iostream_gets() fill the supplied buffer with read data until either a '\n' is found or the supplied buffer is filled just like fgets(). Change-Id: If18b3f6ee500e22f0633a68779ed09f7e0f305ed	7 years ago
Joshua Colp	62afa54977	Merge "res_fax: Handle fax gateway being started more than once." into 16	7 years ago
Joshua Colp	ad37ab9a8f	Merge "res_pjsip_transport_websocket: Properly set src_name for IPv6" into 16	7 years ago
Richard Mudgett	4dd8b5bbb4	res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch ASTERISK-27988 Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843	7 years ago
Rodrigo Ramírez Norambuena	1edd9eb309	make config: os-release output error. Fix not show the error "/bin/sh: /etc/os-release: No such file or directory" when the command 'make config' is run in a System without systemv. The instruction 'make config' pre execute the syntax "$(shell . /etc/os-release && echo $$ID)" to identified if system is a Slackware and Opensuse. This change prevent show the message and is send to the /dev/null Change-Id: I7f43e281a8d9405b2519fc653de82d9b8b645fdf	7 years ago
Joshua Colp	6f27ad59f5	Merge "Create --disable-binary-modules option." into 16	7 years ago
Joshua Colp	390d0b42ca	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
George Joseph	a52b56b4d1	Merge "alembic: increase uri column size" into 16	7 years ago
Sean Bright	245fb462d6	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
Corey Farrell	1b1f47bef6	Create --disable-binary-modules option. This new option can be passed for ./configure or ./tests/CI/buildAsterisk.sh to prevent download/install of binary modules. Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES will result in binary modules being enabled even if the build target is incompatible with those modules. This includes CI scripts which enable categories before disabling specific modules. If more binary modules are offered in the future this will help avoid accidentally downloading them if unwanted or incompatible. Adding a binary module will only require creating a new menuselect entry similar to the existing ones, it will not be necessary to modify the CI scripts. Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166	7 years ago
neutrino88	aa2755cbb3	res/res_rtp_asterisk: remove debug traces generated by an empty frame The realtime text timer pops regularly and sends text frames even if the buffer is empty. This causes a lot of unecessary debug logging. * Made red_write() test if we need to send a frame before calling ast_rtp_write() ASTERISK-28002 Reported by: Emmanuel BUU Tested by: Emmanuel BUU Change-Id: Icf81310c3b8080b615a42060afc02ab41f9523dd	7 years ago
Joshua Colp	3d495f89bc	Merge "pbx_dundi: Added IPv6 support for dundi" into 16	7 years ago
Jenkins2	14c84efbfe	Merge "chan_sip: improved ip:port finding of peers for non-UDP transports." into 16	7 years ago
Jenkins2	777d125516	Merge "sample_configs: noload res_hep.so by default" into 16	7 years ago
Jaco Kroon	46442aa9e5	chan_sip: improved ip:port finding of peers for non-UDP transports. Also remove function peer_ipcmp_cb since it's not used (according to rmudgett). Prior to `b2c4e8660a` (ASTERISK_27457) insecure=port was the defacto standard. That commit also prevented insecure=port from being applied for sip/tcp or sip/tls. Into consideration there are three sets of behaviour: 1. "previous" - before the above commit. 2. "current" - post above commit, pre this one. 3. "new" - post this commit. The problem that the above commit tried to address was guests over TCP. It succeeded in doing that but broke transport!=udp with host!=dynamic. This commit attempts to restore sane behaviour with respect to transport!=udp for host!=dynamic whilst still retaining the guest users over tcp. It should be noted that when looking for a peer, two passes are made, the first pass doesn't have SIP_INSECURE_PORT set for the searched-for peer, thus looking for full matches (IP + Port), the second pass sets SIP_INSECURE_PORT, thus expecting matches on IP only where the matched peer allows for that (in the author's opinion: UDP with insecure=port, or any TCP based, non-dynamic host). In previous behaviour there was special handling for transport=tcp\|tls whereby a peer would match during the first pass if the utilized transport was TCP\|TLS (and the peer allowed that specific transport). This behaviour was wrong, or dubious at best. Consider two dynamic tcp peers, both registering from the same IP (NAT), in this case either peer could match for connections from an IP. It's also this behaviour that prevented SIP guests over tcp. The above referenced commit removed this behaviour, but kept applying the SIP_INSECURE_PORT only to WS\|WSS\|UDP. Since WS and WSS is also TCP based, the logic here should fall into the TCP category. This patch updates things such that the previously non-explicit (TCP behaviour) transport test gets performed explicitly (ie, matched peer must allow for the used transport), as well as the indeterministic source-port nature of the TCP protocol is taken into account. The new match algorithm now looks like: 1. As per previous behaviour, IP address is matched first. 2. Explicit filter with respect to transport protocol, previous behaviour was semi-implied in the test for TCP pure IP match - this now made explicit. 3. During first pass (without SIP_INSECURE_PORT), always match on port. 4. If doing UDP, match if matched against peer also has SIP_INSECURE_PORT, else don't match. 5. Match if not a dynamic host (for non-UDP protocols) 6. Don't match if this is WS\|WSS, or we can't trust the Contact address (presumably due to NAT) 7. Match (we have a valid Contact thus if the IP matches we have no choice, this will likely only apply to non-NAT). To logic-test this we need a few different scenarios. Towards this end, I work with a set number of peers defined in sip.conf: [peer1] host=1.1.1.1 transport=tcp [peer2] host=1.1.1.1 transport=udp [peer3] host=1.1.1.1 port=5061 insecure=port transport=udp [peer4] host=1.1.1.2 transport=udp,tcp [peer5] host=dynamic transport=udp,tcp Test cases for UDP: 1 - incoming UDP request from 1.1.1.1: - previous: - pass 1: * peer1 or peer2 if from port 5060 (indeterminate, depends on peer ordering) * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 - current: as per previous. - new: - pass 1: * peer2 if from port 5060 * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 2 - incoming UDP request from 1.1.1.2: - previous: - pass 1: * peer5 if registered from 1.1.1.2 and port matches * peer4 if source port is 5060 - pass 2: * no match (guest) - current: as previous. - new as previous (with the variation that if peer5 didn't have udp as allowed transport it would not match peer5 whereas previous and current code could). 3 - incoming UDP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address and source port matches. - pass 2: * peer5 if insecure=port is additionally set. * no match (guest) - current - as per previous - new - as per previous Test cases for TCP based transports: 4 - incoming TCP request from 1.1.1.1 - previous: - pass 1 (indeterministic, depends on ordering of peers in memory): * peer1; or * peer5 if peer5 registered from 1.1.1.1 (irrespective of source port); or * peer2 if the source port happens to be 5060; or * peer3 if the source port happens to be 5061. - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer1 or peer2 if from source port 5060 * peer3 if from source port 5060 * peer5 if registered as 1.1.1.1 and source port matches - pass 2: * no match (guest) - new: - pass 1: * peer 1 if from port 5060 * peer 5 if registered and source port matches - pass 2: * peer 1 5 - incoming TCP request from 1.1.1.2 - previous (indeterminate, depends on ordering): - pass 1: * peer4; or * peer5 if peer5 registered from 1.1.1.2 - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * no match (guest). - new: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * peer4 6 - incoming TCP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer5 if registered from that address and port matches. - pass 2: * no match (guest) - new: as per current. It should be noted the test cases don't make explicit mention of TLS, WS or WSS. WS and WSS previously followed UDP semantics, they will now enforce source port matching. TLS follow TCP semantics. The previous commit specifically tried to address test-case 6, but broke test-cases 4 and 5 in the process. ASTERISK-27881 #close Change-Id: I61a9804e4feba9c7224c481f7a10bf7eb7c7f2a2	7 years ago
Jaco Kroon	d84de695ed	AMI: be less verbose when adding HTTP headers to AMI/HTTP messages. All HTTP/AMI message headers are being sent to the verbose channel. There are multiple places this is happening. Consolidate the loop into a function. Drop the debug/verbose message. Convert to using ast_asprintf to perform the length calculation, memory allocation and snprintf all in one step. Change-Id: Ic45e673fde05bd544be95ad5cdbc69518207c1a1	7 years ago
Matthew Fredrickson	4188e7d6dd	sample_configs: noload res_hep.so by default Change disables loading of res_hep.so in default installation. Loading res_hep has a performance impact whether it's used or not. This disables loading of it in sample config files. Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0 (cherry picked from commit `c8bacd45f1`)	7 years ago
Florian Floimair	595e358761	alembic: increase uri column size When mobile SIP clients register with Asterisk that use some sort of push notifications, the URI can get quite lengthy due to the additional push-service annotations (things like tokens, pn-type, etc.) contained in it. ASTERISK-28022 #close Change-Id: I6c55013bafe79f7e7a1fb6722d2558f553709f2e	7 years ago
Joshua Colp	bd650b6a49	Merge "app_queue: Silence GCC 8 compiler warning" into 16	7 years ago
Joshua Colp	378964f403	Merge "res_pjsip: Reduce processing when a Contact is updated." into 16	7 years ago
Sean Bright	4b88cb383d	app_queue: Silence GCC 8 compiler warning I'm only seeing an error in 14+, so I assume it is due to different compiler options: app_queue.c: In function ‘handle_queue_add_member’: app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11 bytes into a region of size 3 [-Werror=format-overflow=] sprintf(num, "%d", state); ^~ app_queue.c:10234:18: note: directive argument in the range [-2147483648, 99] sprintf(num, "%d", state); ^~~~ Compiler: gcc version 8.0.1 20180414 (experimental) [trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2) Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10	7 years ago
Joshua Colp	6b527d11ae	Merge "pbx_dundi.c: Handle thread shutdown better." into 16	7 years ago
Joshua Colp	b0db0df7b8	Merge "AMI: Remove docs for nonexistent AMI ContactStatus event headers" into 16	7 years ago
Joshua Colp	2454c1d310	Merge "pbx_dundi: Fix debug frame decode string." into 16	7 years ago
Joshua Colp	b880b50922	Merge "pbx_dundi.c: Misc memory management fixes when destroying peers" into 16	7 years ago
Richard Mudgett	fb2f242c49	AMI: Remove docs for nonexistent AMI ContactStatus event headers Change-Id: I5736965c64c44338f7330e85a24bb46818607f19	7 years ago
George Joseph	5645dc6894	Merge "pbx_dundi: Update sample config documentation." into 16	7 years ago
George Joseph	b523aaf699	Merge "res_sorcery_realtime.c: Fix unqualified fetch warning." into 16	7 years ago
George Joseph	0fe2eadbc3	Merge "res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response." into 16	7 years ago
Joshua Colp	b9cd4c6d92	res_pjsip: Reduce processing when a Contact is updated. When a Contact is updated the only material change that qualify support cares about is the underlying configuration for the AOR. In this case we will update things with the new AOR information but otherwise the callback to indicate the Contact has changed can be ignored. This is because it is only when a Contact is added or deleted that material changes occur within the qualify support. An update can't change the URI since it would result in a new Contact so it can be ignored. Change-Id: I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d	7 years ago
Richard Mudgett	236826a111	res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response. We were still getting crashes after the first fix. Somehow we receive a non-2xx final response before we get a 200 final response. With the failure response we had already cleaned up and destroyed some data structures. When the unexpected 200 response comes in we crash. * Add protection code to prevent processing another final T.38 reINVITE response. ASTERISK-27944 Change-Id: I8b5baba8d07fe4d63f0d7d05d3eb9a3d27d40a74	7 years ago
Richard Mudgett	19298141cf	res_sorcery_realtime.c: Fix unqualified fetch warning. The allow_unqualified_fetch option for the sorcery realtime backend blocked actually fetching all rows when the option is set to warn. * Made issue a warning and actually do the request when allow_unqualified_fetch=warn is set. Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312	7 years ago
Kirsty Tyerman	f6c1d94d91	pbx_dundi: Added IPv6 support for dundi Change includes move to netsock2 library. ASTERISK-27164 Reported-by: Adam Secombe Change-Id: Ia9e8dc3d153de7a291dbda4bd87fc827dd2bb846	7 years ago
Richard Mudgett	4e43823d19	pbx_dundi.c: Misc memory management fixes when destroying peers * In destroy_peer(), fixed memory leaks of lookup history strings and qualify transactions when destroying peers. * In destroy_peer(), fixed leaving the registerexpire scheduled callback active when a peer is destroyed on a reload. The reload marks and sweeps peers so any peers not explicitly configured get destroyed. Peers created dynamically from the '' peer will not exist until they re-register after the reload. These destroyed peers caused memory corruption when the registerexpire timer expired. Made build_peer() not schedule any callbacks on the '' peer (empty_eid). It is a special peer that is cloned to dynamically created peers so it doesn't actually get involved in any message transactions. Made do_register_expire() remove the dundi/dpeers AstDB entry when a peer registration expires. * Fix deep_copy_peer() to not copy some things that cannot be copied to the cloned peer structure. Timers, message transactions, and lookup history are specific to a peer instance. * Made set_config() lock around processing the mappings configuration. * Reordered unload_module() to handle load_module() declining the load due to error. Change-Id: Ib846b2b60d027f3a2c2b3b563d9a83a357dce1d6	7 years ago
Richard Mudgett	857926d197	pbx_dundi.c: Handle thread shutdown better. Change-Id: Id52f99bd6a948fe6dd82acc0a28b2447a224fe87	7 years ago
Richard Mudgett	111e73678e	pbx_dundi: Fix debug frame decode string. * Fixed a typo in the name of the REGREQ frame decode string array. * Fixed off by one range check indexing into the frame decode string array. * Removed some unneeded casts associated with the decode string array. Change-Id: I77435e81cd284bab6209d545919bf236ad7933c2	7 years ago
Richard Mudgett	a7ace944ae	pbx_dundi: Update sample config documentation. Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849	7 years ago
Richard Mudgett	0874d5b316	res_rtp_asterisk.c: Fix unused variable warnings Compiling without SRTP support installed resulted in some unused variable warnings. These warnings also showed that the srtp variable was obtained and passed around some functions but not really used even when a system has SRTP installed. Change-Id: I6daad34be3e89b19adef6e2fbe738018975155fc	7 years ago
Joshua Colp	7d8e2389d6	Merge "res_resolver_unbound: Fix leak of config nameserver strings." into 16	7 years ago
Joshua Colp	14da6be84d	Merge "res_pjsip: Resolve transport management leak at shutdown." into 16	7 years ago
Kevin Harwell	80a331d96b	Merge "res_odbc: Allow unload at shutdown." into 16	7 years ago

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