asterisk

Commit Graph

Author	SHA1	Message	Date
Alexander Traud	a4b105c99b	res_rtp_asterisk: In Developer Mode, do not require OpenSSL. OpenSSL is an optional external library and should stay optional even when Developer Mode is configured. ASTERISK-27990 Change-Id: Ia68a4cd5474b26d45e0f43b04032ad598022853b	7 years ago
Joshua Colp	ef647cc31e	Merge "res_rtp_asterisk: Avoid merging command and regular T.140 text packets" into 13	7 years ago
Joshua Colp	cdf6e93b6d	Merge "res_pjsip: Change log message from error to warning for valid use cases" into 13	7 years ago
George Joseph	54364318b5	Merge "res_pjsip: Update default keepalive interval to 90 seconds." into 13	7 years ago
Florian Floimair	fd1b8c57e9	res_pjsip: Change log message from error to warning for valid use cases If a SIP MESSAGE is triggered for an endpoint that is currently not registered - and therefore has no valid contact associated - an error message was logged. Since this is a valid request in a valid use cases this is now changed to a warning, as discussed with Matt Fredrickson on the asterisk-dev mailing list. Change-Id: I55eb62d2712818a58c7532119dec288bd98cf0c0	7 years ago
neutrino88	e5721b01f6	res_rtp_asterisk: Avoid merging command and regular T.140 text packets When realtime text packets are to be sent, the text is accumulated in a buffer and sent regularly by a timer. It can happen that commands such as a backspace, CR, or LF get merged with regular text. This breaks some UAs. The proposed change: * We test if the current packet contains a command. If so we send the buffer immediately. * We test if the buffer contained a command. If so we send the buffer immediately. * We accumulate the text (or the command) in the buffer. ASTERISK-27970 Change-Id: Ifbe993311410fa855cb8aa4a12084db75f413462	7 years ago
Joshua Colp	18e9acc3c1	res_pjsip: Update default keepalive interval to 90 seconds. A change recently went in which disabled the built-in PJSIP keepalive. This defaulted to 90 seconds and kept TCP/TLS connections alive. Disabling this functionality has resulted in a behavior change of not doing keepalives by default resulting in TCP/TLS connections dropping for some people. This change makes our default keepalive interval 90 seconds to match the previous behavior and preserve it. ASTERISK-27978 Change-Id: Ibd9a45f3cbe5d9bb6d2161268696645ff781b1d6	7 years ago
Richard Mudgett	0ade9df3b6	res_pjsip: Update endpoint transport option documentation. Change-Id: I5394fdff6a296efc8e1695a156e616acd932ae52	7 years ago
George Joseph	f4ddc56b9a	Merge "res_pjsip_sdp_rtp: include ice in ANSWER only if offered" into 13	7 years ago
Torrey Searle	bd36ec69e2	res_pjsip_sdp_rtp: include ice in ANSWER only if offered Keep track if ICE candidates were in the SDP offer & only put them in the corresponding SDP answer if the offer condaind ICE candidates ASTERISK-27957 #close Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92	7 years ago
Joshua Colp	03c1bbffd2	res_sorcery_config: Allow configuration section to be used based on name. A problem I've seen countless times is a global or system section for PJSIP not getting applied. This is inevitably the result of the "type=" line missing. This change alleviates that problem. The ability to specify an explicit section name has been added to res_sorcery_config. If the configured section name matches this and there are no unknown things configured the section is taken as being for the given type. Both the PJSIP "global" and "system" types now support this so you can just name your section "global" or "system" and it will be matched and used, even without a "type=" line. ASTERISK-27972 Change-Id: Ie22723663c1ddd24f869af8c9b4c1b59e2476893	7 years ago
Richard Mudgett	33668ab46a	res_config_ldap.c: Fix mem leak in CLI "realtime show ldap status" Change-Id: Ib2a3622b297b0363c62d23958f7a20039bcb9b4c	7 years ago
Joshua Colp	3fe275ca02	Merge "res_pjsip_t38.c: Be smarter about how we respond when T.38 is disabled." into 13	7 years ago
Jenkins2	50e46e3bdb	Merge "res_pjsip_pubsub: segfault in function publish_expire" into 13	7 years ago
Jenkins2	ba5668bb1f	Merge "res_pjsip/pjsip_transport_management.c: Fix deadlock with transport keep alive." into 13	7 years ago
Joshua Colp	6b7a4f07ed	Merge "res_pjsip_t38: Decline T.38 stream on failure case." into 13	7 years ago
Jenkins2	0c7cc3a16b	Merge "res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint" into 13	7 years ago
Alexei Gradinari	9e615dac63	res_pjsip_pubsub: segfault in function publish_expire The function pubsub_on_rx_publish_request incorrectly uses of AST_SCHED_REPLACE_UNREF. The AST_SCHED_REPLACE_UNREF should unref old '_data'. Because of this, there may be a double unref of variable 'publication' when ast_sched_del is unsuccessful that leads to use after free of the 'publication' in publish_expire. ASTERISK-27956 #close Change-Id: Ie0f0cfc7e036953d890b188656010b325a5cdc82	7 years ago
George Joseph	3470409dd6	res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint A new option 'suppress_q850_reason_headers' has been added to the endpoint object. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. The default value is 'no'. ASTERISK-27949 Reported-by: Ross Beer Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1	7 years ago
Joshua Colp	76f2699d2d	res_pjsip_t38: Decline T.38 stream on failure case. When negotiating an incoming T.38 stream the code incorrectly returned failure instead of a decline for the stream when a problem occurred or the configuration didn't allow it. This resulted in SDP offers being rejected with a 488 response in all cases, even when another valid stream was present. This change makes it so the stream is now declined. If no streams are accepted a 488 response is sent while if at least one stream is accepted all the declined streams are, well, declined. ASTERISK-27763 Change-Id: I88bcf793788c412a9839d111a5c736bf6867807c	7 years ago
Richard Mudgett	a82f9762fd	res_pjsip_t38.c: Be smarter about how we respond when T.38 is disabled. We were blindly responding with AST_T38_REFUSED when ANY T.38 control frame came accross the bridge. This causes T.38 Gateway to get confused and the T.38 session to get in a strange state. * Made the T.38 framehook only respond to request frames and ignore response frames. ASTERISK-27657 ASTERISK-27080 Change-Id: I5fb5967c7d1efb30a7ff375f82887ca82a55b05b	7 years ago
Jenkins2	b6a13f0fc2	Merge "res_pjsip_t38.c: Fix crash by ignoring 1xx messages." into 13	7 years ago
Richard Mudgett	b6b49071af	res_pjsip/pjsip_transport_management.c: Fix deadlock with transport keep alive. Using the keep_alive_interval option can result in a deadlock between the pjproject transport manager group lock and the monitored transports ao2 container lock. The pjproject transport manager group lock has to be superior in the locking order to the monitored transports ao2 container lock because of pjproject callbacks called when already holding the group lock. The lock inversion happens when Asterisk attempts to send a keep alive packet over the reliable transports. * Made keepalive_transport_thread() iterate over the monitored transports container rather than use the ao2_callback() method. This avoids holding the container lock when sending the keep alive packet. ASTERISK-26686 Change-Id: I5d5392a52e698bbe41a93f7d8e92bf0e61fe3951	7 years ago
Jenkins2	43dc9c3d5f	Merge "pjsip: Clarify certificate configuration for Websocket." into 13	7 years ago
Joshua Colp	62859ad526	pjsip: Clarify certificate configuration for Websocket. The Websocket transport uses the built-in HTTP server. As a result the TLS configuration is done in http.conf and not in pjsip.conf. This change adds a warning if this is configured in pjsip.conf and also clarifies in the sample configuration file. Change-Id: I187d994d328c3ed274b6754fd4c2a4955bdc6dd9	7 years ago
Richard Mudgett	140ce56f50	res_pjsip_t38.c: Fix crash by ignoring 1xx messages. If we initiated a T.38 reINVITE, we would crash if we received any other 1xx response message except 100 if it were followed by a 200 response. * Made ignore any 1xx response so we do not close out the T.38 negotiation too early. For good measure we'll now accept any 2xx response as acceptance of the reINVITE T.38 offer. NOTE: In v13 we don't actually crash but we should deal with the non-final response messages and not switch to rejected and then to enabled when the 200 comes in. ASTERISK-27944 Change-Id: I0ca88aae708d091db7335af73f41035a212adff4	7 years ago
Joshua Colp	5f3869608a	res_pjsip_pubsub: Hold module reference for publications. Incoming publications need to ensure that the module remains loaded for the lifetime of them. This is now done by holding a reference to the module while the publication exists. This mirrors that of inbound subscriptions. ASTERISK-27783 Change-Id: Ia98c95a15e11af25728d5fb3e56e12cda0cfc7c0	7 years ago
Jenkins2	6b6fa461d6	Merge "res_pjsip_session: Add ability to accept multiple sdp answers" into 13	7 years ago
Jenkins2	a5754e7904	Merge "res_pjsip_messaging: Allow application/* for in-dialog MESSAGEs" into 13	7 years ago
George Joseph	6087c1e575	res_pjsip_messaging: Allow application/* for in-dialog MESSAGEs In addition to text/* content types, incoming_in_dialog_request now accepts application/* content types. Also fixed a length issue when copying the body text. It was one character short. ASTERISK-27942 Change-Id: I4e54d8cc6158dc47eb8fdd6ba0108c6fd53f2818	7 years ago
George Joseph	06966e91fe	res_pjsip_session: Add ability to accept multiple sdp answers pjproject by default currently will follow media forked during an INVITE on outbound calls if the To tag is different on a subsequent response as that on an earlier response. We handle this correctly. There have been reported cases where the To tag is the same but we still need to follow the media. The pjproject patch in this commit adds the capability to sip_inv and also adds the capability to control it at runtime. The original "different tag" behavior was always controllable at runtime but we never did anything with it and left it to default to TRUE. So, along with the pjproject patch, this commit adds options to both the system and endpoint objects to control the two behaviors, and a small logic change to session_inv_on_media_update in res_pjsip_session to control the behavior at the endpoint level. The default behavior for "different tags" remains the same at TRUE and the default for "same tag" is FALSE. Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 ASTERISK-27936 Reported-by: Ross Beer	7 years ago
Jenkins2	4603e24609	Merge "VECTOR: Passing parameters with side effects to macros is dangerous." into 13	7 years ago
Richard Mudgett	cb9475eb3d	VECTOR: Passing parameters with side effects to macros is dangerous. * Fix several instances where we were bumping a ref in the parameter and then unrefing the object if it failed. The way the AST_VECTOR_APPEND() and AST_VECTOR_REPLACE() macros are implemented means if it fails the new value was never evaluated. Change-Id: I2847872a455b11ea7e5b7ce697c0a455a1d0ac9a	7 years ago
Alexander Traud	c8966b8c29	res_http_post: Enable GMime in Solaris 11. Change-Id: Ie434541f18f894c751d2e44bcb3efb3cac626019	7 years ago
Richard Mudgett	99f439dc01	ARI POST DTMF: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e	7 years ago
Alexander Traud	e3de4bc46e	res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP. Previously, Asterisk used its script ./configure, to test whether OpenSSL was built with no-srtp (or was simply too old). However, the header file <openssl/opensslconf.h> is the preferred way to detect the local configuration of OpenSSL. As a positive side-effect the script ./configure does not interleave the detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of individual features of OpenSSL anymore. Change-Id: I3c77c7b00b2ffa2e935632097fa057b9fdf480c0	7 years ago
Jenkins2	49c4c8af98	Merge "res_rtp_asterisk: Allow OpenSSL configured with no-deprecated." into 13	7 years ago
Joshua Colp	2b25fdb80c	Merge "res_crypto: Allow OpenSSL configured with no-deprecated." into 13	7 years ago
Jenkins2	e46ab4743b	Merge "res_srtp: Repair ./configure --with-ssl=PATH." into 13	7 years ago
Richard Mudgett	034a04af2a	AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses. When endpoint specific ACL rules block a SIP request they respond with a 403 forbidden. However, if an endpoint is not identified then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints. * Made endpoint specific ACL rules now respond with a 401 unauthorized which is the same as if an endpoint were not identified. The fix is accomplished by replacing the found endpoint with the artificial endpoint which always fails authentication. ASTERISK-27818 Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32	7 years ago
Alexander Traud	ef2386fcd7	res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. Furthermore, allow OpenSSL configured with no-dh. Additionally, this change allows auto-negotiation of the elliptic curve/group for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. ASTERISK-27910 Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537	7 years ago
Alexander Traud	7763aa824e	res_crypto: Allow OpenSSL configured with no-deprecated. The header <openssl/rsa.h> had to be included explicitly. ASTERISK-27906 Change-Id: I41743801eed998c039d73db7a0762d104a4f75b2	7 years ago
Alexander Traud	e62e069705	res_srtp: Repair ./configure --with-ssl=PATH. ASTERISK-27905 Change-Id: Ibb7dc148a0048f4f9c3b12937ba4240dff0d15e2	7 years ago
Joshua Colp	19f8f85689	Merge "pjsip_options: handle modification of qualify options in realtime" into 13	7 years ago
George Joseph	8b6c84787c	Merge "pjsip_options: show/reload AOR qualify options using CLI" into 13	7 years ago
Alexei Gradinari	9ad3918acd	pjsip_options: handle modification of qualify options in realtime Currentrly pjsip_options code does not handle the situation when the qualify options were changed in realtime database. Only 'module reload res_pjsip' helps. This patch add a check on contact add/update observers if the contact qualify options are different than local aor qualify options. If the qualify options were modified then synchronize the pjsip_options AOR local state. ASTERISK-27872 Change-Id: Id55210a18e62ed5d35a88e408d5fe84a3c513c62	7 years ago
Joshua Colp	7f318c3ab5	Merge "res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change" into 13	7 years ago
Torrey Searle	4b7fd478d5	res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change Certain race conditions between changing bridge types and DTMF can cause the current FLAG_NEED_MARKER_BIT to send the marker bit before the actual first packet of native bridging. This logic keeps track of the ssrc the bridge is currently sending and will correctly ensure the marker bit is set if SSRC as changed from the previous sent packet. ASTERISK-27845 Change-Id: I01858bd0235f1e5e629e20de71b422b16f55759b	7 years ago
Alexei Gradinari	a576f50362	pjsip_options: show/reload AOR qualify options using CLI Currentrly pjsip_options code does not handle the situation when the AOR qualify options were changed. Also there is no way to find out what qualify options are using. This patch add CLI commands to show and synchronize Aor qualify options: pjsip show qualify endpoint <id> Show the current qualify options for all Aors on the PJSIP endpoint. pjsip show qualify aor <id> Show the PJSIP Aor current qualify options. pjsip reload qualify endpoint <id> Synchronize the qualify options for all Aors on the PJSIP endpoint. pjsip reload qualify aor <id> Synchronize the PJSIP Aor qualify options. ASTERISK-27872 Change-Id: I1746d10ef2b7954f2293f2e606cdd7428068c38c	7 years ago
Jenkins2	9564fc19f5	Merge "res_pjsip_endpoint_identifier_ip: Unregister the module for headers." into 13	7 years ago

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4338 Commits (a4b105c99b5a7963be5286f03eaa7925e77c7fc2)