3491 Commits (c5b9ed20fd36f0941954eae2d032bc762e032f6a)

Author	SHA1	Message	Date
George Joseph	c5b9ed20fd	res_pjsip_session: Add cleanup to ast_sip_session_terminate ... If you use ast_request to create a PJSIP channel but then hang it up without causing a transaction to be sent, the session will never be destroyed. This is due ot the fact that it's pjproject that triggers the session cleanup when the transaction ends. app_chanisavail was doing this to get more granular channel state and it's also possible for this to happen via ARI. * ast_sip_session_terminate was modified to explicitly call the cleanup tasks and unreference session if the invite state is NULL AND invite_tsx is NULL (meaning we never sent a transaction). * chan_pjsip/hangup was modified to bump session before it calls ast_sip_session_terminate to insure that session stays valid while it does its own cleanup. * Added test events to session_destructor for a future testsuite test. ASTERISK-26908 #close Reported-by: Richard Mudgett Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9	9 years ago
Sean Bright	cea3742c54	core: Use eventfd for alert pipes on Linux when possible ... The primary win of switching to eventfd when possible is that it only uses a single file descriptor while pipe() will use two. This means for each bridge channel we're reducing the number of required file descriptors by 1, and - if you're using timerfd - we also now have 1 less file descriptor per Asterisk channel. The API is not ideal (passing int arrays), but this is the cleanest approach I could come up with to maintain API/ABI. I've also removed what I believe to be an erroneous code block that checked the non-blocking flag on the pipe ends for each read. If the file descriptor is 'losing' its non-blocking mode, it is because of a bug somewhere else in our code. In my testing I haven't seen any measurable difference in performance. Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d	9 years ago
Joshua Colp	5e2a8ef94a	Merge "modules: change module LOAD_FAILUREs to LOAD_DECLINES" into 13	9 years ago
George Joseph	f882ca2572	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
Torrey Searle	d8967ff2c0	strings.h: Avoid overflows in the string hash functions ... On 2's compliment machines abs(INT_MIN) behavior is undefined and results in a negative value still being returnd. This results in negative hash codes that can result in crashes. ASTERISK-26528 #close Change-Id: Idff550145ca2133792a61a2e212b4a3e82c6517b	9 years ago
George Joseph	4e6e069491	pjproject_bundled: Add 3 upstream patches ... 0035-r5572-svn-backport-dialog-transaction-deadlock.patch 0036-r5573-svn-backport-ua-pjsua-transaction-deadlock.patch 0037-r5576-svn-backport-session-timer-crash.patch Also removed the progress bar from wget download to stdout. ASTERISK-26905 #close Reported-by: Ross Beer Change-Id: I268fb3cf71a3bb24283ff0d24bd8b03239d81256	9 years ago
Corey Farrell	c701550803	Forward declare 'struct ast_json' in asterisk.h ... The ast_json structure is used in many Asterisk headers and is often the only part of json.h used. This adds a forward declaration to asterisk.h and removes the include of json.h from many headers. The declaration has been left in endpoints.h and stasis.h to avoid problems with source files that use ast_json functions without directly including json.h. ari.h continues to include json.h as it uses enum ast_json_encoding_format. Change-Id: Id766aabce6bed56626d27e8d29f559b5e687b769	9 years ago
zuul	f040eb0502	Merge "CEL: Remove header declarations of non-existant functions." into 13	9 years ago
Sean Bright	c9648f4690	astobj2: Prevent potential deadlocks with ao2_global_obj_release ... The ao2_global_obj_release() function holds an exclusive lock on the global object while it is being dereferenced. Any destructors that run during this time that call ao2_global_obj_ref() will deadlock because a read lock is required. Instead, we make the global object inaccessible inside of the write lock and only dereference it once we have released the lock. This allows the affected destructors to fail gracefully. While this doesn't completely solve the referenced issue (the error message about not being able to create an IQ continues to be shown) it does solve the backtrace spew that accompanied it. ASTERISK-21009 #close Reported by: Marcello Ceschia Change-Id: Idf40ae136b5070dba22cb576ea8414fbc9939385	9 years ago
Corey Farrell	3c23ebdef4	CEL: Remove header declarations of non-existant functions. ... ast_cel_alloc and ast_cel_destroy do not exist in code, remove them from the headers. Change-Id: I99ce848e2e109e7d61771559f559b9e57973e45c	9 years ago
Sean Bright	79a2c26c03	core: Remove embedded module support ... This has not worked for some time and is no longer actively maintained. Change-Id: I5110b0db69c152761b58fa025cb0a53b0e544d99	9 years ago
Kevin Harwell	f1b34e6eb4	AMI: Updated version ... Updated the AMI version for the following reason (see CHANGES for more details): The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now contains a new optional parameter, 'MatchHeader'. Change-Id: I9aeac4decc89f9b464b3f026e97c7ef1acc79242	9 years ago
zuul	b79e67ba47	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support." into 13	9 years ago
Richard Begg	398e5ec16c	res_pjsip_session: Enable RFC3578 overlap dialing support. ... Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	9 years ago
zuul	d7ba743329	Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references." into 13	9 years ago
Sean Bright	218f618095	res_hep: Capture actual transport type in use ... Rather than hard-coding UDP, allow consumers of the HEP API to specify which protocol is in use. Update the PJSIP provider to pass in the current protocol type. ASTERISK-26850 #close Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978	9 years ago
Sean Bright	38cebc73a3	thread safety: Don't use getprotobyname() ... POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48	9 years ago
George Joseph	9b756662a8	res_pjsip: Symmetric transports ... A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	9 years ago
Richard Mudgett	adad6020be	autochan/mixmonitor/chanspy: Fix unsafe channel locking and references. ... Dereferencing struct ast_autochan.chan without first calling ast_autochan_channel_lock() is unsafe because the pointer could change at any time due to a masquerade. Unfortunately, ast_autochan_channel_lock() itself uses struct ast_autochan.chan unsafely and can result in a deadlock if the original channel happens to get destroyed after a masquerade in addition to the pointer getting changed. The problem is more likely to happen with v11 and earlier because masquerades are used to optimize out local channels on those versions. However, it could still happen on newer versions if the channel is executing a dialplan application when the channel is transferred or redirected. In this situation a masquerade still must be used. * Added a lock to struct ast_autochan to safely be able to use ast_autochan.chan while trying to get the channel lock in ast_autochan_channel_lock(). The locking order is the channel lock then the autochan lock. Locking in the other direction requires deadlock avoidance. * Fix unsafe ast_autochan.chan usages in app_mixmonitor.c. * Fix unsafe ast_autochan.chan usages in app_chanspy.c. * app_chanspy.c: Removed unused autochan parameter from next_channel(). ASTERISK-26867 Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592	9 years ago
Mark Michelson	7bc69753bc	Add rtcp-mux support ... This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	9 years ago
Joshua Colp	eac818801b	bridge_native_rtp: Handle case where channel joins already suspended. ... The bridge_native_rtp module did not properly handle the case where a smart bridge operation occurs while a channel is suspended. In this scenario the module would incorrectly set up local or remote RTP bridging despite the media having to flow through Asterisk. The remote endpoint would see two media streams and experience wonky audio. The module has been changed so that it ensures both channels are not suspended when performing the native RTP bridging and this requirement has been documented in the bridge technology. ASTERISK-26781 Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c	9 years ago
zuul	a3584c6834	Merge "res_pjsip: Record the serializer earlier on the tdata." into 13	9 years ago
Sean Bright	5eb7875243	realtime: Centralize some common realtime backend code ... All of the realtime backends create artificial ast_categorys to pass back into the core as query results. These categories have no filename or line number information associated with them and the backends differ slightly on how they create them. So create a couple helper macros to help make things more consistent. Also updated the call sites to remove redundant error messages about memory allocation failure. Note that res_config_ldap sets the category filename to the 'table name' but that is not read by anything in the core, so I've dropped it. Change-Id: I3a1fd91e0c807dea1ce3b643b0a6fe5be9002897	9 years ago
Richard Mudgett	eb9ae4f7cb	res_pjsip: Record the serializer earlier on the tdata. ... When PJPROJECT needs to do a DNS resolution and there is not a cached entry available, the SIP request message goes out on the PJSIP monitor thread instead of the original serializer thread. Thus when the response comes back it does not get processed by the original sending serializer. This patch records the serializer on tdata before passing a request message to PJPROJECT where it can in Asterisk code. There are several places in PJPROJECT for outbound registration and publishing support that would need to record the serializer. Unfortunately, without replacing the PJPROJECT DNS resolver as was done in v14 we cannot fix those without modifying PJPROJECT. Even if we backported the DNS resolver from v14, the outbound registration refresh timer does not go out on a serializer thread but the PJSIP monitor thread. Fortunately, Asterisk's outbound publish support doesn't use the auto refresh timer that would also not go out under the serializer thread. This patch is v13 only. ASTERISK-26669 ASTERISK-26738 Change-Id: I9997b9ed6dbcebd2c37d6a67dc6dcee9c78914a4	9 years ago
George Joseph	be77b845d9	res_pjsip_pubsub: Correctly implement persisted subscriptions ... This patch fixes 2 original issues and more that those 2 exposed. * When we send a NOTIFY, and the client either doesn't respond or responds with a non OK, pjproject only calls our pubsub_on_evsub_state callback, no others. Since pubsub_on_evsub_state (which does the sub_tree cleanup) does not expect to be called back without the other callbacks being called first, it just returns leaving the sub_tree orphaned. Now pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE which is what pjproject will set to tell us that it was the transaction that timed out or failed and not the subscription itself timing our or being terminated by the client. If is TSX_STATE, pubsub_on_evsub_state now does the proper cleanup regardless of the state of the subscription. * When a client renews a subscription, we don't update the persisted subscription with the new expires timestamp. This causes subscription_persistence_recreate to prune the subscription if/when asterisk restarts. Now, pubsub_on_rx_refresh calls subscription_persistence_update to apply the new expires timestamp. This exposed other issues however... * When creating a dialog from rdata (which sub_persistence_recreate does from the packet buffer) there must NOT be a tag on the To header (which there will be when a client refreshes a subscription). If there is one, pjsip_dlg_create_uas will fail. To address this, subscription_persistence_update now accepts a flag that indicates that the original packet buffer must not be updated. New subscribes don't set the flag and renews do. This makes sure that when the rdata is recreated on asterisk startup, it's done from the original subscribe packet which won't have the tag on To. * When creating a dialog from rdata, we were setting the dialog's remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq. When the client tried to resubscribe after a restart with the correct cseq, we'd reject the request with an Invalid CSeq error. * The acts of creating a dialog and evsub by themselves when recreating a subscription does NOT restart pjproject's subscription timer. The result was that even if we did correctly recreate the subscription, we never removed it if the client happened to go away or send a non-OK response to a NOTIFY. However, there is no pjproject function exposed to just set the timer on an evsub that wasn't created by an incoming subscribe request. To address this, we create our own timer using ast_sip_schedule_task. This timer is used only for re-establishing subscriptions after a restart. An earlier approach was to add support for setting pjproject's timer (via a pjproject patch) and while that patch is still included here, we don't use that call at the moment. While addressing these issues, additional debugging was added and some existing messages made more useful. A few formatting changes were also made to 'pjsip show scheduled tasks' to make displaying the subscription timers a little more friendly. ASTERISK-26696 ASTERISK-26756 Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e	9 years ago
Sean Bright	8936568515	manager: Restore Originate failure behavior from Asterisk 11 ... In Asterisk 11, if the 'Originate' AMI command failed to connect the provided Channel while in extension mode, a 'failed' extension would be looked up and run. This was, I believe, unintentionally removed in 51b6c49. This patch restores that behavior. This also adds an enum for the various 'synchronous' modes in an attempt to make them meaningful. ASTERISK-26115 #close Reported by: Nasir Iqbal Change-Id: I8afbd06725e99610e02adb529137d4800c05345d	9 years ago
Richard Mudgett	a199f94908	res_agi: Prevent an AGI from eating frames it should not. (Re-do) ... A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43	10 years ago
Richard Mudgett	6bed318a66	Frame deferral: Revert API refactoring. ... There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496	10 years ago
George Joseph	cb504ad6ae	Merge "ari: Implement 'debug all' and request/response logging" into 13	10 years ago
George Joseph	b39c04fc02	Merge "PJPROJECT logging: Fix detection of max supported log level." into 13	10 years ago
Richard Mudgett	1484a991e1	Add notes about embedded ast_frame structs holding a format ref. ... mod_format.h: Note ast_filestream.fr holds a format ref. translate.h: Note ast_trans_pvt.f holds a format ref. Change-Id: I86bda354d725207b41e08920355d7c31b2d7f749	10 years ago
George Joseph	17f4989d49	ari: Implement 'debug all' and request/response logging ... The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf	10 years ago
Richard Mudgett	30cb4eb57f	PJPROJECT logging: Fix detection of max supported log level. ... The mechanism used for detecting the maximum log level compiled into the linked pjproject did not work. The API call simply stores the requested level into an integer and does no range checking. Asterisk was assuming that there was range checking and limited the new value to the allowable range. To get the actual maximum log level compiled into the linked pjproject we need to get and save off the initial set log level from pjproject. This is the maximum log level supported. * Get and save off the initial log level setting before altering it to the desired level on startup. This has to be done by a macro rather than calling a core function to avoid incorrectly linking pjproject. * Split the initial log level warning messages to warn if the linked pjproject cannot support the requested startup level and if it is too low to get the pjproject buildopts for "pjproject show buildopts". * Adjust the CLI "pjproject set log level" to check the saved max log level and to generate normal output messages instead of a warning message. ASTERISK-26743 #close Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4	10 years ago
Kevin Harwell	883e7fde31	abstract/fixed/adpative jitter buffer: disallow frame re-inserts ... It was possible for a frame to be re-inserted into a jitter buffer after it had been removed from it. A case when this happened was if a frame was read out of the jitterbuffer, passed to the translation core, and then multiple frames were returned from said translation core. Upon multiple frames being returned the first is passed on, but sebsequently "chained" frames are put back into the read queue. Thus it was possible for a frame to go back into the jitter buffer where this would cause problems. This patch adds a flag to frames that are inserted into the channel's read queue after translation. The abstract jitter buffer code then checks for this flag and ignores any frames marked as such. Change-Id: I276c44edc9dcff61e606242f71274265c7779587	10 years ago
Aaron An	0047b1bc49	res_rtp_asterisk: Fix bug in function CHANNEL(rtcp, all_rtt) ... Function CHANNEL(rtcp,all_rtt) CHANNEL(rtcp,all_loss) CHANNEL(rtcp,all_jitter) always return 0.0 due to wrong define of macro "AST_RTP_SATA_SET" and "AST_RTP_STAT_STRCPY". It should compare "combined" with "stat" not "current_stat". ASTERISK-26710 #close Reported-by: Aaron An Tested-by: AaronAn Change-Id: Id4140fafbf92e2db689dac5b17d9caa009028a15	10 years ago
Richard Mudgett	75a6afbec5	MESSAGE: Flush Message/ast_msg_queue channel alert pipe. ... ASTERISK-25083 Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2	10 years ago
George Joseph	ebc67d3053	res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command ... The PJSIPShowRegistrationsInbound AMI command was just dumping out all AORs which was pretty useless and resource heavy since it had to get all endpoints, then all aors for each endpoint, then all contacts for each aor. PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail events which meets the intended purpose of the other command and has significantly less overhead. Also, some additional fields that were added to Contact since the original creation of the ContactStatusDetail event have been added to the end of the event. For compatibility purposes, PJSIPShowRegistrationsInbound is left intact. ASTERISK-26644 #close Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a	10 years ago
Joshua Colp	8d56016ae4	Merge "tcptls: Use new certificate upon sip reload" into 13	10 years ago
Joshua Colp	28b76ed667	Merge "PJPROJECT logging: Made easier to get available logging levels." into 13	10 years ago
Guido Falsi	2ceb609edb	res_rtp: Fix regression when IPv6 is not available. ... The latest Release candidate fails to create RTP streams when IPv6 is not available. Due to the changes made in September the ast_sockaddr structure passed around to create these streams is always of AF_INET6 type, causing failure when used for IPv4. This patch adds a utility function to check for availability of IPv6 and applies such check at startup to determine how to create the ast_sockaddr structures. ASTERISK-26617 #close Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e	10 years ago
Richard Mudgett	44fe4a5769	PJPROJECT logging: Made easier to get available logging levels. ... Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	10 years ago
zuul	eec82c6221	Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no" into 13	10 years ago
Michael Kuron	8e77d6f520	tcptls: Use new certificate upon sip reload ... Previously, a TLS server socket would only be restarted upon sip reload if the bind address had changed. This commit adds checking for changes to TLS parameters like certificate, ciphers, etc. so they get picked up without requiring a reload of the entire chan_sip module. This does not affect open connections in any way, but new connections will use the new TLS parameters. The changes also apply to HTTP and Manager. ASTERISK-26604 #close Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6	10 years ago
George Joseph	425da14927	build: Backport addition of librt check to configure.ac ... A while back, a master-only change was made to check for librt which should probably have been cherry-picked to 13 at that time. Sometime between then and now, part of that change did make it into 13 but it was incomplete and non-functional. This patch backports the rest of the librt check and allows the link of libasteriskpj to use the results. Change-Id: I1424008fd8c90f389dda53162ec4a340b253a3c1	10 years ago
Mark Michelson	bde3d022a3	manager: update minor version ... Based on bridge video AMI event changes, bump the minor version of AMI. Change-Id: I02586bd6cafc0baa33ea98c2f75356c0f5e03435	10 years ago
zuul	3135a745e3	Merge "res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak." into 13	10 years ago
George Joseph	404596b790	channel: Fix issues in hangup scenarios caused by frame deferral ... ASTERISK-26343 Change-Id: I06dbf7366e26028251964143454a77d017bb61c8	10 years ago
zuul	b745c326c2	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source" into 13	10 years ago
zuul	f68790d46a	Merge "Revert "Revert "Add API for channel frame deferral.""" into 13	10 years ago
Joshua Colp	4c3d25875f	Merge "Add X.509 subject alternative name support to TLS certificate verification." into 13	10 years ago