asterisk

Commit Graph

Author	SHA1	Message	Date
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
Alexander Traud	b615df06d3	res_pjsip_endpoint_identifier_ip: Unregister the module for headers. Asterisk uses Reference Counting to track whether a module can be unloaded. Every consumer who requires a module, increases the reference count. When the consumer goes, is unloaded itself, it has to decrease the reference count on all its used/required modules. That way core stop gracefully works on the command-line interface (CLI): One module after the other is unloaded. A recent change broke this for the module res_pjsip. ASTERISK-27861 Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3	7 years ago
Alexander Traud	99b24dc63b	res_pjsip: Register pjsip_transport_management not externally but internally. The module (res_)pjsip_transport_management got moved into res_pjsip. It is no longer an independent/external module with (un)load_module and therefore has to register just internally with res_pjsip. ASTERISK-27860 Change-Id: Icd0413be7d2e98b92f51e6d6c353f2570bb4be95	7 years ago
Jenkins2	118eef8907	Merge "Fix GCC 8 build issues." into 13	7 years ago
Joshua Colp	ac9d6b0523	Merge "pjsip: Rewrite OPTIONS support with new eyes." into 13	7 years ago
Corey Farrell	d893e57c90	Fix GCC 8 build issues. This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	7 years ago
Matthew Fredrickson	316efcddb9	res_hep: Adds hostname resolution support for capture_address Previously, only an IP address would be accepted for the capture_address config setting in hep.conf. This change allows capture_address to be a resolvable hostname or an IP address. ASTERISK-27796 #close Reported-By: Sebastian Gutierrez Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f	7 years ago
Jenkins2	2d0fb07f53	Merge "res_ari: Remove requirement that body exists when debug is on." into 13	7 years ago
Jenkins2	8966a39e9d	Merge "res_pjsip/pjsip_distributor.c: Add missing off-nominal request response." into 13	7 years ago
Joshua Colp	6192e40e04	Merge "pjsip: Increase maximum number of usable ciphers & other cleanups" into 13	7 years ago
Joshua Colp	a36d6e3674	res_ari: Remove requirement that body exists when debug is on. The "ari set debug" code for incoming requests incorrectly assumed that all requests would contain a body. If one did not exist the request would be incorrectly rejected. The response that was sent was also incomplete as an incorrect function was used to construct the response. The code has now been changed to no longer require a request to have a body and the response updated to use the correct function. ASTERISK-27801 Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8	7 years ago
Sean Bright	850e554793	pjsip: Increase maximum number of usable ciphers & other cleanups * Increase maximum number of ciphers from 100 to 256 (or whatever PJ_SSL_SOCK_MAX_CIPHERS is #define'd to) * Simplify logic in cipher_name_to_id() * Make signed/unsigned comparison consistent Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412 Reported by: Ondřej Holas Change-Id: Iea620f03915a1b873e79743154255c3148a514e7	7 years ago
Richard Mudgett	51a6a986f1	res_pjsip/pjsip_distributor.c: Add missing off-nominal request response. Change-Id: I389579b39c523d1d1e8ce020ef549a8bb5781c9b	7 years ago
Richard Mudgett	3663968705	res_pjsip/pjsip_distributor.c: Pull some assignments out of if tests. Change-Id: I3d30d638b53a4bbe9bf9aad853c649d583894112	7 years ago
George Joseph	42abc9c430	Merge "BuildSystem: Add DragonFly BSD." into 13	7 years ago
Joshua Colp	bea52b3706	pjsip: Rewrite OPTIONS support with new eyes. The OPTIONS support in PJSIP has organically grown, like many things in Asterisk. It has been tweaked, changed, and adapted based on situations run into. Unfortunately this has taken its toll. Configuration file based objects have poor performance and even dynamic ones aren't that great. This change scraps the existing code and starts fresh with new eyes. It leverages all of the APIs made available such as sorcery observers and serializers to provide a better implementation. 1. The state of contacts, AORs, and endpoints relevant to the qualify process is maintained. This state can be updated by external forces (such as a device registering/unregistering) and also the reload process. This state also includes the association between endpoints and AORs. 2. AORs are scheduled and not contacts. This reduces the amount of work spent juggling scheduled items. 3. Manipulation of which AORs are being qualified and the endpoint states all occur within a serializer to reduce the conflict that can occur with multiple threads attempting to modify things. 4. Operations regarding an AOR use a serializer specific to that AOR. 5. AORs and endpoint state act as state compositors. They take input from lower level objects (contacts feed AORs, AORs feed endpoint state) and determine if a sufficient enough change has occurred to be fed further up the chain. 6. Realtime is supported by using observers to know when a contact has been registered. If state does not exist for the associated AOR then it is retrieved and becomes active as appropriate. The end result of all of this is best shown with a configuration file of 3000 endpoints each with an AOR that has a static contact. In the old code it would take over a minute to load and use all 8 of my cores. This new code takes 2-3 seconds and barely touches the CPU even while dealing with all of the OPTIONS requests. ASTERISK-26806 Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082	7 years ago
George Joseph	c5d19565fe	Merge "bridge_softmix: Forward TEXT frames" into 13	7 years ago
Joshua Colp	cb4e991181	Merge "res_pjsip: Fix initialization of extended stringfields." into 13	7 years ago
Alexander Traud	7b219311eb	BuildSystem: Add DragonFly BSD. ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	7 years ago
Jenkins2	b3cec9d445	Merge "res_pjsip: Fix deadlock on reliable transport shutdown." into 13	7 years ago
Corey Farrell	ea14243971	res_pjsip: Fix initialization of extended stringfields. It is possible for initialization of extended stringfields to fail. Add checks for this failure. Change-Id: I062e09852db3d37ceefaf6c2048958fa0118304f	7 years ago
Jenkins2	e516c34c02	Merge "res_musiconhold: Don't restart MOH from beginning after announcement." into 13	7 years ago
George Joseph	be7d4faed5	bridge_softmix: Forward TEXT frames Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	7 years ago
Jenkins2	ad0ba520b5	Merge "pjsip_scheduler.c: Add ability to trace scheduled tasks." into 13	7 years ago
Jenkins2	fb9cbd524b	Merge "res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations." into 13	7 years ago
Jenkins2	f83f5c48b2	Merge "pjsip_scheduler.c: Fix some corner cases." into 13	7 years ago
Ben Ford	f2e140e8a7	res_musiconhold: Don't restart MOH from beginning after announcement. This reverts a problem introduced by the fix for ASTERISK_24329. Now, when an announcement is played while waiting in a queue, music on hold will not restart from the beginning of the sound file and will instead pick up where it left off. However, the incorrect behavior in ASTERISK_24329 is now present again; if an announcement X seconds long is played when music on hold starts, music on hold will start X seconds into the file. ASTERISK-27774 #close Reported by: lvl Change-Id: I86b2885ee7063268f9b9747eddb788336ade989b	7 years ago
Richard Mudgett	b92ebdba5f	pjsip_scheduler.c: Add ability to trace scheduled tasks. When a scheduled task is created you can pass in the AST_SIP_SCHED_TASK_TRACK flag. This new flag causes scheduling events to be logged. Change-Id: I91967eb3d5a220915ce86881a28af772f9a7f56b	7 years ago
Richard Mudgett	12aa25b2e1	res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3	7 years ago
Richard Mudgett	dfdc9a2575	pjsip_scheduler.c: Fix some corner cases. * Fix the periodic interval wander because it may take significant time between the sched thread queueing the task in the serializer and the serializer actually executing the task. The time it takes to actually execute the task was already taken into account. * Pass a schtd ref to the serializer when we queue a scheduled task on the serializer. We don't want it going away on us while it is in the serializer queue. * Skip the scheduled task if the task was canceled between queueing the task to the serializer and the serializer actually executing the task. * Reorder struct ast_sip_sched_task to avoid unnecessary padding. Removed task_id and added next_periodic. * Hold a ref to the passed in serializer so the serializer cannot go away on the scheduled task. ASTERISK_26806 Change-Id: I6c8046b75f6953792c8c30e55b836a4291143f24	7 years ago
Richard Mudgett	af36823907	pjsip_scheduler.c: Sort "pjsip show scheduled_tasks" output. * A side benefit is that the scheduled tasks are not completely blocked while the CLI command executes. * Adjusted the "Task Name" column width to have more room for longer names. Change-Id: Iec64aa463ee8b10eef90120e00c38b1fb444087e	7 years ago
Jenkins2	264af6bec0	Merge "res_pjsip_notify.c: enable in-dialog NOTIFY" into 13	7 years ago
Jenkins2	dfd0529abc	Merge "pjsip_scheduler.c: Fix ao2 usage errors." into 13	7 years ago
Chris-Savinovich	724d926d46	res_pjsip_session: Rewrite o= with external_media_address. It now appends the external IP address on the o= line of the SDP packet. The decision was made to write the numeric IP address as opposed to the RFC that states the FQDN should be used if and when available. We believe the usage of literal IP address will help avoid potential problems. ASTERISK-27614 #close Change-Id: I84f3360f3606b8c4e8d161edb228799ec0b8a302	7 years ago
Nathan Bruning	6a4afe09ce	res_pjsip_notify.c: enable in-dialog NOTIFY This patch adds support to send in-dialog SIP NOTIFY commands on chan_pjsip channels, similar to the functionality recently added for chan_sip (ASTERISK_27461). This extends res_pjsip_notify to allow for in-dialog messages. ASTERISK-27697 Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29	7 years ago
Richard Mudgett	c4f02c975b	pjsip_scheduler.c: Fix ao2 usage errors. * Removed several invalid uses of OBJ_NOLOCK. These uses resulted in the 'tasks' container being accessed without a lock in a multi-threaded environment. A recipe for crashes. * Removed needlessly obtaining schtd object references. If the caller providing you a pointer to an object doesn't have a valid reference then you cannot safely get one from it. * Getting a ref to 'tasks' when you aren't copying the pointer into another location is useless. The 'tasks' container pointer is global. * Removed many unnecessary uses of RAII_VAR. * Make ast_sip_schedule_task() name parameter const. ASTERISK_26806 Change-Id: I5c62488e651314e2a1dbc01f5b078a15512d73db	7 years ago

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4303 Commits (e3de4bc46e0332050f3fd296202abe94d13c20a5)