136 Commits (d35c494df1bfc1cc44d7483d9750f31d0184d020)

Author	SHA1	Message	Date
George Joseph	d2a1457e0b	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	9 years ago
Joshua Colp	1cddd954f9	Merge "res_pjsip: Refactor load_module/unload_module" into 13	9 years ago
zuul	953ba9da88	Merge "res_pjsip: Handle pjsip_dlg_create_uas deprecation" into 13	9 years ago
George Joseph	b1b797e0e7	res_pjsip: Refactor load_module/unload_module ... load_module was just too hairy with every step having to clean up all previous steps on failure. Some of the pjproject init calls have now been moved to a separate load_pjsip function and the unload_pjsip function was enhanced to clean up everything if an error happened at any stage of the load process. In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns and ast_threadpool_shutdowns were also corrected. Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302	9 years ago
George Joseph	c1bf014ea0	res_pjsip: Handle pjsip_dlg_create_uas deprecation ... Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically increments the lock on the returned dialog. To account for this, configure.ac now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use the original call or the new one. If the new one was used, the ref count is decremented before returning. ASTERISK-25751 #close Reported-by Josh Colp Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8	9 years ago
George Joseph	2451d4e455	res_pjsip: Fix infinite recursion when loading transports from realtime ... Attempting to load a transport from realtime was forcing asterisk into an infinite recursion loop. The first thing transport_apply did was to do a sorcery retrieve by id for an existing transport of the same name. For files, this just returns the previous object from res_sorcery_config's internal container, if any. For realtime, the res_sourcery_realtime driver looks in the database and finds the existing row but now it has to rehydrate it into a sorcery object which means calling... transport_apply. And so it goes. The main issue with loading from realtime (apart from the loop) was that transport stores structures and pointers directly in the ast_sip_transport structure instead of the separate ast_transport_state structure. This patch separates those items into the ast_sip_transport_state structure. The pattern is roughly the same as res_pjsip_outbound_registration. Although all current usages of ast_sip_transport and ast_sip_transport_state were modified to use the new ast_sip_get_transport_state API, the original items are left in ast_sip_transport and kept updated to maintain ABI compatability for third-party modules. They are marked as deprecated and noted that they're now in ast_sip_transport_state. ASTERISK-25606 #close Reported-by: Martin Moučka Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19	9 years ago
Richard Mudgett	f87c3275cc	res_pjsip: Add CLI "pjsip dump endpt [details]" ... Dump the res_pjsip endpt internals. In non-developer mode we will not document or make easily accessible the "details" option even though it is still available. The user has to know it exists to use it. Presumably they would also be aware of the potential crash warning below. Warning: PJPROJECT documents that the function used by this CLI command may cause a crash when asking for details because it tries to access all active memory pools. Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb	10 years ago
Joshua Colp	236896f391	Merge "pjsip: Add option global/regcontext" into 13	10 years ago
Joshua Colp	092c0db493	Merge "pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address" into 13	10 years ago
George Joseph	219c204a41	pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address ... On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88	10 years ago
Daniel Journo	22801a06ee	pjsip: Add option global/regcontext ... Added new global option (regcontext) to pjsip. When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given endpoint who registers or unregisters with us. ASTERISK-25670 #close Reported-by: Daniel Journo Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62	10 years ago
Richard Mudgett	cf8e7a580b	res_pjsip: Create human friendly serializer names. ... PJSIP name formats: pjsip/aor/<aor>-<seq> -- registrar thread pool serializer pjsip/default-<seq> -- default thread pool serializer pjsip/messaging -- messaging thread pool serializer pjsip/outreg/<registration>-<seq> -- outbound registration thread pool serializer pjsip/pubsub/<endpoint>-<seq> -- pubsub thread pool serializer pjsip/refer/<endpoint>-<seq> -- REFER thread pool serializer pjsip/session/<endpoint>-<seq> -- session thread pool serializer pjsip/websocket-<seq> -- websocket thread pool serializer Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084	10 years ago
Matt Jordan	f62b642fe3	res/res_pjsip: Fix off nominal crash with requests that fail and have a timer ... When a request is sent using pjsip_endpt_send_request and fails, a condition exists where the request wrapper, which is an AO2 object, may be de-ref'd more times than it should. This occurs when the request's callback is called, and, in the callback, the timer on the PJSIP heap is cancelled. When that occurs, the request wrapper's lifetime is decremented. When pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of the request wrapper again, even though we've already cancelled the reference associated with the timer. This patch checks the return result of pj_timer_heap_cancel_if_active before removing the reference associated with the timer. We now only decrement it in this case if a timer is cancelled as a result of the function call. Change-Id: I21332343a1a019c1117076f9bf2df27be2850102	10 years ago
Mark Michelson	2f9cb7d62b	res_pjsip: Deny requests when threadpool queue is backed up. ... We have observed situations where the SIP threadpool may become deadlocked. However, because incoming traffic is still arriving, the SIP threadpool's queue can continue to grow, eventually running the system out of memory. This change makes it so that incoming traffic gets rejected with a 503 response if the queue is backed up too much. Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816	10 years ago
Joshua Colp	f2725c8b77	res_pjsip: Move URI validation to use time. ... In a realtime based system with a limited number of threadpool threads it is possible for a deadlock to occur. This happens when permanent endpoint state is updated, which will cause database queries to be done. These queries may result in URI validation being done which is done synchronously using a PJSIP thread. If all PJSIP threads are in use processing traffic they themselves may be blocked waiting to get the permanent endpoint container lock when identifying an endpoint. This change moves URI validation to occur at use time instead of configuration time. While this comes at a cost of not seeing a problem until you use it it does solve the underlying deadlock problem. ASTERISK-25486 #close Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a	10 years ago
Richard Mudgett	3329c714f7	res_pjsip: Fix deadlock when sending out-of-dialog requests. ... The struct send_request_wrapper has a pjsip lock associated with it that is created non-recursive. There is a code path for the struct send_request_wrapper lock that will attempt to lock it recursively. The reporter's deadlock showed that the thread calling endpt_send_request() deadlocked itself right after the wrapper object got created. Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited MWI NOTIFY messages can hit this deadlock. * Replaced the struct send_request_wrapper pjsip lock with the mutex lock that can come with an ao2 object since all of Asterisk's mutexes are recursive. Benefits include removal of code maintaining the pjsip non-recursive lock since ao2 objects already know how to maintain their own lock and the lock will show up in the CLI "core show locks" output. ASTERISK-25435 #close Reported by: Dmitriy Serov Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d	10 years ago
Mark Michelson	9d1f176e29	res_pjsip: Copy default_from_user to avoid crash. ... The default_from_user retrieval function was pulling the default_from_user from the global configuration struct in an unsafe way. If using a database as a backend configuration store, the global configuration struct is short-lived, so grabbing a pointer from it results in referencing freed memory. The fix here is to copy the default_from_user value out of the global configuration struct. Thanks go to John Hardin for discovering this problem and proposing the patch on which this fix is based. ASTERISK-25390 #close Reported by Mark Michelson Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c	10 years ago
Mark Michelson	ac62928d6b	res_pjsip: Change default from user value. ... When Asterisk sends an outbound SIP request, if there is no direct reason to place a specific value for the username in the From header, Asterisk would generate a UUID. For example, this would happen when sending outbound OPTIONS requests when qualifying or when sending outbound INVITE requests when originating (if no explicit caller ID were provided). The issue is that some SIP providers reject these sorts of requests with a "Name too long" error response. This patch aims to fix this by changing the default outbound username in From headers to "asterisk". This value can be overridden by changing the default_from_user option in the global options if desired. ASTERISK-25377 #close Reported by Mark Michelson Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190	10 years ago
Mark Michelson	ad9cb6c2ce	res_pjsip: Fix contact refleak on stateful responses. ... When sending a stateful response, creation of the transaction can fail, most commonly because we are trying to create a transaction from a retransmitted request. When creation of the transaction fails, we end up leaking a reference to a contact that was bumped when the response was created. This patch adds the missing deref and fixes the reference leak. Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07	10 years ago
Joshua Colp	2a4eee0cd9	res_pjsip: Add common ast_sip_get_host_ip API. ... Modules commonly used the pj_gethostip function for retrieving the IP address of the host. This function does not cache the result and may result in a DNS lookup occurring, or additional work. If the DNS server is unreachable or network issues arise this can cause the pj_gethostip function to block for a period of time. This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string function which does the same thing but caches the host IP address at module load time. This results in no additional work being done each time the local host IP address is needed. ASTERISK-25342 #close Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e	10 years ago
Richard Mudgett	c126afe18f	res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message. ... If the saved SUBSCRIBE message is not parseable for whatever reason then Asterisk could crash when libpjsip tries to parse the message and adds an error message to the parse error list. * Made ast_sip_create_rdata() initialize the parse error rdata list. The list is checked after parsing to see that it remains empty for the function to return successful. ASTERISK-25306 Reported by Mark Michelson Change-Id: Ie0677f69f707503b1a37df18723bd59418085256	10 years ago
Joshua Colp	8521a86367	res_pjsip: Ensure sanitized XML is NULL terminated. ... The ast_sip_sanitize_xml function is used to sanitize a string for placement into XML. This is done by examining an input string and then appending values to an output buffer. The function used by its implementation, strncat, has specific behavior that was not taken into account. If the size of the input string exceeded the available output buffer size it was possible for the sanitization function to write past the output buffer itself causing a crash. The crash would either occur because it was writing into memory it shouldn't be or because the resulting string was not NULL terminated. This change keeps count of how much remaining space is available in the output buffer for text and only allows strncat to use that amount. Since this was exposed by the res_pjsip_pidf_digium_body_supplement module attempting to send a large message the maximum allowed message size has also been increased in it. A unit test has also been added which confirms that the ast_sip_sanitize_xml function is providing NULL terminated output even when the input length exceeds the output buffer size. ASTERISK-25304 #close Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302	10 years ago
Joshua Colp	2749721791	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	10 years ago
Mark Michelson	d9094ddd73	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b	10 years ago
Richard Mudgett	ada7346792	res_pjsip: Need to use the same serializer for a pjproject SIP transaction. ... All send/receive processing for a SIP transaction needs to be done under the same threadpool serializer to prevent reentrancy problems inside pjproject and res_pjsip. * Add threadpool API call to get the current serializer associated with the worker thread. * Pick a serializer from a pool of default serializers if the caller of res_pjsip.c:ast_sip_push_task() does not provide one. This is a simple way to ensure that all outgoing SIP request messages are processed under a serializer. Otherwise, any place where a pushed task is done that would result in an outgoing out-of-dialog request would need to be modified to supply a serializer. Serializers from the default serializer pool are picked in a round robin sequence for simplicity. A side effect is that the default serializer pool will limit the growth of the thread pool from random tasks. This is not necessarily a bad thing. * Made pjsip_distributor.c save the thread's serializer name on the outgoing request tdata struct so the response can be processed under the same serializer. This is a cherry-pick from master. **** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a NOTE: session_inv_on_state_changed() is disassociating the dialog from the session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED. Unfortunately this is a tad too soon because our BYE request transaction has not completed yet. ASTERISK-25183 #close Reported by: Matt Jordan Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a	10 years ago
Matt Jordan	0a1700d286	Merge "threadpool, res_pjsip: Add serializer group shutdown API calls." into 13	10 years ago
Mark Michelson	028fa54620	res_pjsip_nat: Rewrite route set when required. ... When performing some provider testing, the rewrite_contact option was interfering with proper construction of a route set when sending an ACK after receiving a 200 OK response to an INVITE. The initial INVITE was sent to address sip:foo. The 200 OK had a Contact header with URI sip:bar. In addition, the 200 OK had Record-Route headers for sip:baz and sip:foo, in that order. Since the Record-Route headers had the lr parameter, the result should have been: * Set R-URI of the ACK to sip:bar. * Add Route headers for sip:foo and sip:baz, in that order. However, the rewrite_contact option resulted in our rewriting the Contact header on the 200 OK to sip:foo. The result was: * R-URI remained sip:foo. * We added Route headers for sip:foo and sip:baz, in that order. The result was that sip:bar was not indicated in the ACK at all, so the far end never received our ACK. The call eventually dropped. The intention of rewrite_contact is to rewrite the most immediate destination of our SIP request to be the same address on which we received a request or response. In the case of processing a SIP response with Record-Route headers, this means that instead of rewriting the Contact header, we should instead rewrite the bottom-most Record-Route header. In the case of processing a SIP request with Record-Route headers, this means we rewrite the top-most Record-route header. Like when we rewrite the Contact header, we also ensure to update the dialog's route set if it exists. ASTERISK-25196 #close Reported by Mark Michelson Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f	10 years ago
Richard Mudgett	84c12f9e0c	threadpool, res_pjsip: Add serializer group shutdown API calls. ... A module trying to unload needs to wait for all serializers it creates and uses to complete processing before unloading. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059	10 years ago
Kevin Harwell	31c77b157b	res_pjsip: Add option to force G.726 to be treated as AAL2 packed. ... Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615	10 years ago
George Joseph	60e2fbfe62	res_pjsip: Refactor endpt_send_transaction (qualify_timeout) ... This patch refactors the transaction timeout processing to eliminate calling the lower level public pjsip functions and reverts to calling pjsip_endpt_send_request again. This is the result of me noticing a possible incompatibility with pjproject-2.4 which was causing contact status flapping. The original version of this feature used the lower level calls to get access to the tsx structure in order to cancel the transaction when our own timer expires. Since we no longer have that access, if our own timer expires before the pjsip timer, we call the callbacks and just let the pjsip transaction take it's own course. When the transaction ends, it discovers the callbacks have already been run and just cleans itself up. A few messages in pjsip_configuration were also added/cleaned up. ASTERISK-25105 #close Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Mark Michelson	e39bd6ba46	res_pjsip_outbound_registration: Don't fail on delayed processing: 13. ... This is the Asterisk 13 version of a change to master that allows for registration responses to be processed successfully potentially after the original transaction has timed out. The main difference between this and the master change is that the master version has API changes that are unacceptable for 13. For 13, this is worked around by adding a new API call that the outbound registration code uses instead. The following is the text from the master version of this commit: Odd behaviors have been observed during outbound registrations. The most common problem witnessed has been one where a request with authentication credentials cannot be created after receiving a 401 response. Other behaviors include apparently processing an incorrect SIP response. Inspecting the code led to an apparent issue with regards to how we handle transactions in outbound registration code. When a response to a REGISTER arrives, we save a pointer to the transaction and then push a task onto the registration serializer. Between the time that we save the pointer and push the task, it's possible for the transaction to be destroyed due to a timeout. It's also possible for the address to be reused by the transaction layer for a new transaction. To allow for authentication of a REGISTER request to be authenticated after the transaction has timed out, we now also hold a reference to the original REGISTER request instead of the transaction. The function for creating a request with authentication has been altered to take the original request instead of the transaction where the original request was sent. ASTERISK-25020 Reported by Mark Michelson Change-Id: If1ee5f601be839479a219424f0358a229f358f7c	10 years ago
George Joseph	f70d21b2cf	res_pjsip: Validate that contact uris start with sip: or sips: ... Currently we use pjsip_parse_hdr to validate contact uris but it appears that it allows uris without a scheme if there's a port supplied. I.E myexample.com will fail but myexample.com:5060 will pass even though it has no scheme. This causes SEGVs later on whenever the uri is used. To prevent this, permanent_contact_validate has been updated to check that the scheme is either 'sip' or 'sips'. 2 uses of possibly-null endpoint have also been fixed in create_out_of_dialog_request. ASTERISK-24999 Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2 Reported-by: Brad Latus	10 years ago
George Joseph	674b18bdf0	pjsip_options: Add qualify_timeout processing and eventing ... This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Matt Jordan	f1abf51b73	Merge "res_pjsip: Refactor endpt_send_request to include transaction timeout" into 13	10 years ago
George Joseph	bf46799f0e	res_pjsip: Refactor endpt_send_request to include transaction timeout ... This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html Since we currently have no control over pjproject transaction timeout, this patch pulls the pjsip_endpt_send_request function out of pjproject and into res_pjsip/endpt_send_transaction in order to implement that capability. Now when the transaction is initiated, we also schedule our own pj_timer with our own desired timeout. If the transaction completes before either timeout, pjproject cancels its timer, and calls our tsx callback where we cancel our timer and run the app callback. If the pjproject timer times out first, pjproject calls our tsx callback where we cancel our timer and run the app callback. If our timer times out first, we terminate the transaction which causes pjproject to cancel its timer and call our tsx callback where we run the app callback. Regardless of the scenario, pjproject is calling the tsx callback inside the group_lock and there are checks in the callback to make sure it doesn't run twice. As part of this patch ast_sip_send_out_of_dialog_request was created to replace its similarly named private function. It takes a new timeout argument in milliseconds (<= 0 to disable the timeout). ASTERISK-24863 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747	10 years ago
George Joseph	1b6f6ff841	res_pjsip: Add global option to limit the maximum time for initial qualifies ... Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Matthew Jordan	88b0fa7755	res_pjsip: Add an 'auto' option for DTMF Mode ... This patch adds support for automatically detecting the type of DTMF that a PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto', the channel created for an endpoint will attempt to determine if RFC 4733 DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type for the channel will be set to inband. Review: https://reviewboard.asterisk.org/r/4438 ASTERISK-24706 #close Reported by: yaron nahum patches: yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	5737650a67	res_pjsip: add CLI command to show global and system configuration ... Added a new CLI command for res_pjsip that shows both global and system configuration settings: pjsip show settings ASTERISK-24918 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/4597/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	2270c40d33	res_pjsip: config option 'timers' can't be set to 'no' ... When setting the configuration option 'timers' equal to 'no' the bit flag was not properly negated. This patch clears all associated flags and only sets the specified one. pjsip will handle any necessary flag combinations. Also went ahead and did similar for the '100rel' option. ASTERISK-24910 #close Reported by: Ray Crumrine Review: https://reviewboard.asterisk.org/r/4582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Mark Michelson	85feac857c	Add stateful PJSIP response API call, and use it for out-of-dialog responses. ... Asterisk had an issue where retransmissions of MESSAGE requests resulted in Asterisk processing the retransmission as if it were a new MESSAGE request. This patch fixes the issue by creating a transaction in PJSIP on the incoming request. This way, if a retransmission arrives, the PJSIP transaction layer will resend the response and Asterisk will not ever see the retransmission. ASTERISK-24920 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4532/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	d0df545a44	res_pjsip: Enable unload of all modules at shutdown. ... * Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes caused by running PJSIP functions from non-PJSIP threads. * Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing crashes in some cases. In theory pj_shutdown() should take care of this. * Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at shutdown. * Resolve leaked config global in res_pjsip_notify. * Unregister pubsub pjsip service module. * Implement cleanup for res_pjsip_session. ASTERISK-24731 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4498/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	b1e9552b08	chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. ... Incoming PJSIP call legs that have not been answered yet send unnecessary "180 Ringing" or "183 Progress" messages every time a connected line update happens. If the outgoing channel is also PJSIP then the incoming channel will always send a "180 Ringing" or "183 Progress" message when the outgoing channel sends the INVITE. Consequences of these unnecessary messages: * The caller can start hearing ringback before the far end even gets the call. * Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of the first Asterisk box. When connected line first went into Asterisk in v1.8, chan_sip received an undocumented option "rpid_immediate" that defaults to disabled. When enabled, the option immediately passes connected line update information to the caller in "180 Ringing" or "183 Progress" messages as described above. * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or "183 Progress" messages. The default is "no" to disable sending the unnecessary messages. ASTERISK-24781 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4473/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	6ca98524bf	Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks. ... Valgrind found some memory leaks associated with ast_pjsip_rdata_get_endpoint(). The leaks would manifest when sending responses to OPTIONS requests, processing MESSAGE requests, and res_pjsip supplements implementing the incoming_request callback. * Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in res/res_pjsip.c:supplement_on_rx_request(), res/res_pjsip/pjsip_options.c:send_options_response(), res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and res/res_pjsip_messaging.c:send_response(). * Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in res/res_pjsip_nat.c:nat_on_rx_message(). * Fixed inconsistent but benign return value in res/res_pjsip/pjsip_options.c:options_on_rx_request(). Review: https://reviewboard.asterisk.org/r/4511/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	94fe4a9178	res_pjsip: Allow configuration of endpoint identifier query order ... Updated some documentation stating that endpoint identifiers registered without a name are place at the front of the lookup list. Also renamed register method 'ast_sip_register_endpoint_identifier_by_name' to 'ast_sip_register_endpoint_identifier_with_name' ASTERISK-24840 Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	1f428f25f0	res_pjsip: Allow configuration of endpoint identifier query order ... This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	522f063186	res_pjsip: Add reason comment. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	0497b7b155	Revert - res_pjsip: Allow configuration of endpoint identifier query order ... Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	110b99646c	res_pjsip: Allow configuration of endpoint identifier query order ... It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	f0d018e249	res_pjsip: Add a log message when creating a UAC dialog to a target URI that is invalid. ... ASTERISK-24499 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	323c0927ac	various: cleanup issues found during leak hunt ... In this collection of small patches to prevent Valgrind errors are: fixes for reference leaks in config hooks, evaluating a parameter beyond bounds, and accessing a structure after a lock where it could have been already free'd. Review: https://reviewboard.asterisk.org/r/4407/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago