116 Commits (03e08d5c33c60b3e099feb322161bec85c157413)

Author	SHA1	Message	Date
Thomas Guebels	03e08d5c33	pjsip_transport_events: Avoid monitor destruction ... When a transport is disconnected, several events can arrive following each other. The first event will be PJSIP_TP_STATE_DISCONNECT and it will trigger the destruction of the transport monitor object. The lookup for the transport monitor to destroy is done using the transport key, that contains the transport destination host:port. A reconnect attempt by pjsip will be triggered as soon something needs to send a packet using that transport. This can happen directly after a disconnect since ca Subsequent events can arrive later like PJSIP_TP_STATE_DESTROY and will also try to trigger the destruction of the transport monitor if not already done. Since the lookup for the transport monitor to destroy is done using the transport key, it can match newly created transports towards the same destination and destroy their monitor object. Because of this, it was sometimes not possible to monitor a transport after one or more disconnections. This fix adds an additional check on the transport pointer to ensure only a monitor for that specific transport is removed. Fixes: #923	12 months ago
George Joseph	1872abe672	security_agreements.c: Refactor the to_str functions and fix a few other bugs ... * A static array of security mechanism type names was created. * ast_sip_str_to_security_mechanism_type() was refactored to do a lookup in the new array instead of using fixed "if/else if" statments. * security_mechanism_to_str() and ast_sip_security_mechanisms_to_str() were refactored to use ast_str instead of a fixed length buffer to store the result. * ast_sip_security_mechanism_type_to_str was removed in favor of just referencing the new type name array. Despite starting with "ast_sip_", it was a static function so removing it doesn't affect ABI. * Speaking of "ast_sip_", several other static functions that started with "ast_sip_" were renamed to avoid confusion about their public availability. * A few VECTOR free loops were replaced with AST_VECTOR_RESET(). * Fixed a meomry leak in pjsip_configuration.c endpoint_destructor caused by not calling ast_sip_security_mechanisms_vector_destroy(). * Fixed a memory leak in res_pjsip_outbound_registration.c add_security_headers() caused by not specifying OBJ_NODATA in an ao2_callback. * Fixed a few ao2_callback return code misuses. Resolves: #845	1 year ago
George Joseph	a45e5ebf8d	security_agreement.c: Always add the Require and Proxy-Require headers ... The `Require: mediasec` and `Proxy-Require: mediasec` headers need to be sent whenever we send `Security-Client` or `Security-Verify` headers but the logic to do that was only in add_security_headers() in res_pjsip_outbound_register. So while we were sending them on REGISTER requests, we weren't sending them on INVITE requests. This commit moves the logic to send the two headers out of res_pjsip_outbound_register:add_security_headers() and into security_agreement:ast_sip_add_security_headers(). This way they're always sent when we send `Security-Client` or `Security-Verify`. Resolves: #789	1 year ago
Maximilian Fridrich	f3cc1e7fbd	res_pjsip: mediasec: Add Security-Client headers after 401 ... When using mediasec, requests sent after a 401 must still contain the Security-Client header according to draft-dawes-sipcore-mediasec-parameter. Resolves: #48	3 years ago
Marcel Wagner	b83af13f65	res_pjsip: Update contact_user to point out default ... Updates the documentation for the 'contact_user' field to point out the default outbound contact if no contact_user is specified 's' ASTERISK-30316 #close Change-Id: I61f24fb9164e4d07e05908a2511805281874c876	3 years ago
George Joseph	7684c9e907	pjsip_transport_events: Fix possible use after free on transport ... It was possible for a module that registered for transport monitor events to pass in a pjsip_transport that had already been freed. This caused pjsip_transport_events to crash when looking up the monitor for the transport. The fix is a two pronged approach. 1. We now increment the reference count on pjsip_transports when we create monitors for them, then decrement the count when the transport is going to be destroyed. 2. There are now APIs to register and unregister monitor callbacks by "transport key" which is a string concatenation of the remote ip address and port. This way the module needing to monitor the transport doesn't have to hold on to the transport object itself to unregister. It just has to save the transport_key. * Added the pjsip_transport reference increment and decrement. * Changed the internal transport monitor container key from the transport->obj_name (which may not be unique anyway) to the transport_key. * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that fills a buffer with the transport_key using a passed-in pjsip_transport. * Added the following functions: ast_sip_transport_monitor_register_key ast_sip_transport_monitor_register_replace_key ast_sip_transport_monitor_unregister_key and marked their non-key counterparts as deprecated. * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use the new "key" monitor functions. NOTE: res_pjsip_registrar also uses the transport monitor functionality but doesn't have a persistent object other than contact to store a transport key. At this time, it continues to use the non-key monitor functions. ASTERISK-30244 Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b	3 years ago
Igor Goncharovsky	096529d33f	res_pjsip_outbound_registration: Allow to use multiple proxies for registration ... Current registration code use pjsip_parse_uri to verify outbound_proxy that is different from the reading this option for the endpoint. This made value with multiple proxies invalid for registration pjsip settings. Removing URI validation helps to use registration through multiple proxies. ASTERISK-30217 #close Change-Id: I064558e66f04b9f3260c46181812a01349761357	3 years ago
Maximilian Fridrich	0d2e140123	res_pjsip: Add mediasec capabilities. ... This patch adds support for mediasec SIP headers and SDP attributes. These are defined in RFC 3329, 3GPP TS 24.229 and draft-dawes-sipcore-mediasec-parameter. The new features are implemented so that a backbone for RFC 3329 is present to streamline future work on RFC 3329. With this patch, Asterisk can communicate with Deutsche Telekom trunks which require these fields. ASTERISK-30032 Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2	3 years ago
Ben Ford	881a3f2306	res_pjsip: Add TEL URI support for basic calls. ... This change allows TEL URI requests to come through for basic calls. The allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To headers will now allow TEL URIs, as well as the request URI. Support is only for TEL URIs present in traffic from a remote party. Asterisk does not generate any TEL URIs on its own. ASTERISK-26894 Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a	3 years ago
Naveen Albert	31dc28ab09	res_pjsip_outbound_registration: Make max random delay configurable. ... Currently, PJSIP will randomly wait up to 10 seconds for each outbound registration's initial attempt. The reason for this is to avoid having all outbound registrations attempt to register simultaneously. This can create limitations with the test suite where we need to be able to receive inbound calls potentially within 10 seconds of starting up. For instance, we might register to another server and then try to receive a call through the registration, but if the registration hasn't happened yet, this will fail, and hence this inconsistent behavior can cause tests to fail. Ultimately, this requires a smaller random value because there may be no good reason to wait for up to 10 seconds in these circumstances. To address this, a new config option is introduced which makes this maximum delay configurable. This allows, for instance, this to be set to a very small value in test systems to ensure that registrations happen immediately without an unnecessary delay, and can be used more generally to control how "tight" the initial outbound registrations are. ASTERISK-29965 #close Change-Id: Iab989a8e94323e645f3a21cbb6082287c7b2f3fd	3 years ago
Naveen Albert	604785f931	res_pjsip_outbound_registration: Show time until expiration ... Adjusts the pjsip show registration(s) commands to show the amount of seconds remaining until a registration expires. ASTERISK-29845 #close Change-Id: Ic4fea15a1a1056c424416def49d1ca8e776c0483	4 years ago
Alexander Traud	a85f2bf34d	res: Fix for Doxygen. ... These are the remaining issues found in /res. ASTERISK-29761 Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d	4 years ago
Alexander Traud	463f6c83e8	res_pjsip: Fix for Doxygen. ... ASTERISK-29747 Change-Id: Ic7a1e9453f805a6264fe86c96b7d18b87b376084	4 years ago
Josh Soref	9ae9893c63	res: Spelling fixes ... Correct typos of the following word families: identifying structures actcount initializer attributes statement enough locking declaration userevent provides unregister session execute searches verification suppressed prepared passwords recipients event because brief unidentified redundancy character the module reload operation backslashes accurate incorrect collision initializing instance interpreted buddies omitted manually requires queries generator scheduler configuration has owner resource performed masquerade apparently routable ASTERISK-29714 Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda	4 years ago
Nick French	8f6e0f9367	res_pjsip: dont return early from registration if init auth fails ... If set_outbound_initial_authentication_credentials() fails, handle_client_registration() bails early without creating or sending a register message. [set_outbound_initial_authentication_credentials() failures can occur during the process of retrieving an oauth access token.] The return from handle_client_registration is ignored, so returning an error doesn't do any good. This is a real problem when the registration request is a re-register, because then the registration will still be marked 'active' despite the re-register never being sent at all. So instead, log a warning but let the registration be created and sent (and probably fail) and follow the normal registration failed retry/abort logic. ASTERISK-29315 #close Change-Id: I2e03b1ea7fba1fa1a8279086aa4b17679e7fa7fa	5 years ago
Ivan Poddubnyi	7d15655f9d	res_pjsip_diversion: Fix adding more than one histinfo to Supported ... New responses sent within a PJSIP sessions are based on those that were sent before. Therefore, adding/modifying a header once causes it to be sent on all responses that follow. Sending 181 Call Is Being Forwarded many times first adds "histinfo" duplicated more and more, and eventually overflows past the array boundary. This commit adds a check preventing adding "histinfo" more than once, and skipping it if there is no more space in the header. Similar overflow situations can also occur in res_pjsip_path and res_pjsip_outbound_registration so those were also modified to check the bounds and suppress duplicate Supported values. ASTERISK-29227 Reported by: Ivan Poddubny Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322	5 years ago
George Joseph	9a4486e9fb	Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff" ... This reverts commit 2fe76dd816. Reason for revert: Too many issues reported. Need to research and correct. ASTERISK-29230 ASTERISK-29231 Reported by: Michael Maier Change-Id: I6453af680e17d8ffe7af2c5de7e1b2a58c8793cb	5 years ago
George Joseph	2fe76dd816	res_pjsip_outbound_registration.c: Use our own scheduler and other stuff ... * Instead of using the pjproject timer heap, we now use our own pjsip_scheduler. This allows us to more easily debug and allows us to see times in "pjsip show/list registrations" as well as being able to see the registrations in "pjsip show scheduled_tasks". * Added the last registration time, registration interval, and the next registration time to the CLI output. * Removed calls to pjsip_regc_info() except where absolutely necessary. Most of the calls were just to get the server and client URIs for log messages so we now just save them on the client_state object when we create it. * Added log messages where needed and updated most of the existong ones to include the registration object name at the start of the message. Change-Id: I4534a0fc78c7cb69f23b7b449dda9748c90daca2	5 years ago
George Joseph	78b01f41ae	res_pjsip_outbound_registration: Fix SRV failover on timeout ... In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06	6 years ago
Kevin Harwell	d5d41409e2	res_pjsip_outbound_registration: add support for SRV failover ... ASTERISK-28624 Change-Id: I8da7c300dd985ab7b10dbd5194aff2f737808561	6 years ago
Joshua Colp	e73eba85c1	res_pjsip_outbound_registration: Extend documentation for "max_retries". ... If the "max_retries" option is set to 0 then upon failure no further attemps are made, so explicitly document the behavior. ASTERISK-28602 Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf	6 years ago
Corey Farrell	021ce938ca	astobj2: Remove legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	3077ad0c24	stasis: Add internal filtering of messages. ... This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 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	7 years ago
Joshua Colp	7b59cfc777	Merge "res_pjsip: Fix deadlock on reliable transport shutdown."	8 years ago
Richard Mudgett	237d341bbd	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	8 years ago
George Joseph	f91263cf46	res_pjsip: Correct usages of pjproject's timer heap ... Fix some timer heap initializations and cancels to try and prevent crashes and timer heap issues. Change-Id: I64885d190fa22097d1b55987091375541e57a7ee	8 years ago
Richard Mudgett	97cc67b12f	res_pjsip: Fix deadlock on reliable transport shutdown. ... A deadlock can happen when the PJSIP monitor thread is shutting down a connection oriented transport (TCP/TLS) used by a subscription at the same time as another thread tries to send something for that subscription. The deadlock is between the pjsip monitor thread attempting to get the dialog lock and another thread sending something for that dialog when it tries to get the transport manager lock. * res_pjsip_pubsub.c: Avoid the deadlock by pushing the subscription removal to the subscription serializer. * res_pjsip_registrar.c: Pushed off incoming registration contact removals to a default serializer as a precaution. Removing the contacts involves sorcery access which in this case will involve database access. Depending upon the setup, the database may not be on the same machine and could take awhile. We don't want to hold up the pjsip monitor thread with potentially long access times. ASTERISK-27706 Change-Id: I56b647aea565f24dba33e9e5ebeed4cd3f31f8c4	8 years ago
George Joseph	2b9aa6b5bb	res_pjsip_pubsub: Prune subs with reliable transports at startup ... In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36	8 years ago
Jenkins2	a231e1d155	Merge "Remove redundant module checks and references."	8 years ago
Corey Farrell	527cf5a570	Remove redundant module checks and references. ... This removes references that are no longer needed due to automatic references created by module dependencies. In addition this removes most calls to ast_module_check as they were checking modules which are listed as dependencies. Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e	8 years ago
Corey Farrell	679fa5fb34	Add missing OPTIONAL_API and ARI dependences. ... I've audited all modules that include any header which includes asterisk/optional_api.h. All modules which use OPTIONAL_API now declare those dependencies in AST_MODULE_INFO using requires or optional_modules as appropriate. In addition ARI dependency declarations have been reworked. Instead of declaring additional required modules in res/ari/resource_*.c we now add them to an optional array "requiresModules" in api-docs for each module. This allows the AST_MODULE_INFO dependencies to include those missing modules. Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606	8 years ago
Corey Farrell	9cfdb81e91	loader: Add dependency fields to module structures. ... * Declare 'requires' and 'enhances' text fields on module info structure. * Rename 'nonoptreq' to 'optional_modules'. * Update doxygen comments. Still need to investigate dependencies among modules I cannot compile. Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf	8 years ago
Corey Farrell	7ef38d399a	res_pjsip_outbound_registration: Fix leak on vector add failure. ... Change-Id: I774b88b3c9da41edd4dc8d78f095481f52f2bd46	8 years ago
Corey Farrell	606ae3484a	Add missing menuselect dependencies. ... This adds menuselect dependencies for modules that use symbols of other modules. ASTERISK-27390 Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385	8 years ago
George Joseph	d178f497d2	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	8 years ago
Richard Mudgett	1bec781cce	res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown. ... The fix for the issue is broken up into three parts. This is part three which handles the client side of REGISTER requests. The registered contact may no longer be valid on the server when the transport used is reliable and the connection is broken. * Re-REGISTER our contact if the reliable transport is broken after registration completes. We attempt to re-REGISTER immediately to minimize the time we are unreachable. Time may have already passed between the connection being broken and the loss being detected. * Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's are still correct if an allocation failure happens. ASTERISK-27147 Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83	8 years ago
Richard Mudgett	615b6a200a	res_pjsip_outbound_registration.c: Misc fixes. ... * Remove unnecessary CMP_STOP. * In handle_client_registration() use DEBUG_ATLEAST() to only do work needed for the debug log message when the debug log message is needed. * In sip_outbound_registration_state_destroy() check state->registration for NULL. Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80	8 years ago
George Joseph	747beb1ed1	modules: change module LOAD_FAILUREs to LOAD_DECLINES ... In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting if a module can't be loaded. If the user wishes to retain the FAILURE behavior for a specific module, they can use the "require" or "preload-require" keyword in modules.conf. A new API was added to logger: ast_is_logger_initialized(). This allows asterisk.c/check_init() to print to the error log once the logger subsystem is ready instead of just to stdout. If something does fail before the logger is initialized, we now print to stderr instead of stdout. Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25	9 years ago
Joshua Colp	4e3b0cedba	res_pjsip_transport_websocket: Add support for IPv6. ... This change adds a PJSIP patch (which has been contributed upstream) to allow the registration of IPv6 transport types. Using this the res_pjsip_transport_websocket module now registers an IPv6 Websocket transport and uses it for the corresponding traffic. ASTERISK-26685 Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647	9 years ago
Joshua Colp	063af910eb	Merge "res_pjsip_outbound_registration: Subscribe to network change events"	9 years ago
George Joseph	22242fef5d	res_pjsip_outbound_registration: Subscribe to network change events ... Outbound registration now subscribes to network change events published by res_stun_monitor and refreshes all registrations when an event happens. The 'pjsip send (un)register' CLI commands were updated to accept '*all' as an argument to operate on all registrations. The 'PJSIP(Un)Register' AMI commands were also updated to accept '*all'. ASTERISK-26808 #close Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25	9 years ago
Joshua Colp	2046743938	config: Improve documentation and behavior of outbound_proxy option. ... This change updates the documentation for the outbound_proxy option to ensure it is consistently stated that a full SIP URI must be provided for the option. The res_pjsip_outbound_registration module has also been changed so that the provided outbound_proxy value is checked to ensure it is a URI and if not an error is output stating so. ASTERISK-26782 Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593	9 years ago
Richard Mudgett	0b660c9989	res_pjsip: Update authentication realm documentation. ... Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. An empty inbound auth realm represents the global section's default_realm value when the authentication object is used to challenge an incoming request. An empty outgoing auth realm is treated as a don't care wildcard when the authentication object is used to respond to an incoming authentication challenge. ASTERISK-26799 Change-Id: Id3952f7cfa1b6683b9954f2c5d2352d2f11059ce	9 years ago
Richard Mudgett	4b3d3fc741	res_pjsip_outbound_registration.c: Filter redundant statsd reporting. ... Increasing the testsuite shutdown timeout before forcibly killing Asterisk allowed more events to be sent out. Some tests failed as a result. The tests/channels/pjsip/statsd/registrations failed because we now get the statsd events that a comment in the test configuration stated couldn't be intercepted. Unfortunately, we get a variable number of events because of internal status state transition races generating redundant statsd events. We were reporting redundant statsd PJSIP.registrations.state changes for internal state changes that equated to the same thing publicly. * Made update_client_state_status() filter out redundant statsd updates. ASTERISK-26527 Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646	9 years ago
Richard Mudgett	bb196323f9	res_pjsip: Fix tdata leaks in off nominal paths. ... Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b	9 years ago
George Joseph	ae81b55361	res_pjsip_outbound_registration: Clean up state when registration is deleted ... Nothing was cleaning up the registration state object when ast_sorcery_delete was called on a registration. So, the registration was deleted from sorcery but the state object went right on refreshing the registration (or failing to refresh the registration) with the peer. * Added a 'deleted' observer on registration that removes the state object. ASTERISK-25964 #close Reported-by Matt Jordan Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23	10 years ago
Sebastian Damm	d14d1ba826	res_pjsip_outbound_registration: generate correct Contact URI for TLS ... There are two types of SIP URIs indicating a secure transport: * sips:user@example.org * sip:user@example.org;transport=tls When using a sips URI, Asterisk checks incoming INVITEs and answers from the other side for sips URIs, and rejects the packet if there are only sip URIs. So Asterisk should only generate a sips Contact URI if the other side supports it. This patch makes Asterisk generate either a sip or sips Contact URI depending on the format of the server URI. If you want a sip URI, use: server_uri=sip:example.org\;transport=tls If you want a sips URI, use: server_uri=sips:example.org ASTERISK-25990 #close Reported-by: Sebastian Damm Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2	10 years ago
Mark Michelson	7b8b6e2e4f	AST-2016-004: Fix crash on REGISTER with long URI. ... Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d	10 years ago
George Joseph	ba8adb4ce3	res_pjsip/config_transport: Allow reloading transports. ... The 'reload' mechanism actually involves closing the underlying socket and calling the appropriate udp, tcp or tls start functions again. Only outbound_registration, pubsub and session needed work to reset the transport before sending requests to insure that the pjsip transport didn't get pulled out from under them. In my testing, no calls were dropped when a transport was changed for any of the 3 transport types even if ip addresses or ports were changed. To be on the safe side however, a new transport option was added (allow_reload) which defaults to 'no'. Unless it's explicitly set to 'yes' for a transport, changes to that transport will be ignored on a reload of res_pjsip. This should preserve the current behavior. Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf	10 years ago