asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	b64f4bb6ee	bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media. This change fixes two issues: 1. During a swap operation bridging added the new channel before having the swap channel leave. This was not handled in bridge_native_rtp and could result in a channel not getting reinvited back to Asterisk. After this change the swap channel will leave first and the new channel will then join. 2. If a re-invite was received after a session had been established any upstream elements (such as bridge_native_rtp) were not notified that they may want to re-evaluate things. After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs and upstream can react. AST-1524 #close Review: https://reviewboard.asterisk.org/r/4378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	7f9b28b0c6	ARI: Improve wiki documentation This patch improves the documentation of ARI on the wiki. Specifically, it addresses the following: * Allowed values and allowed ranges weren't documented. This was particularly frustrating, as Asterisk would reject query parameters with disallowed values - but we didn't tell anyone what the allowed values were. * The /play/id operation on /channels and /bridges failed to document all of the added media resource types. * Documentation for creating a channel into a Stasis application failed to note when it occurred, and that creating a channel into Stasis conflicts with creating a channel into the dialplan. * Some other minor tweaks in the mustache templates, including italicizing the parameter type, putting the default value on its own sub-bullet, and some other nicities. Review: https://reviewboard.asterisk.org/r/4351 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	ceedd40370	res_parking: Fix crash due to race condition when unloading. There is currently a race condition when unloading the res_parking module. Depending on the will of the universe the subscription invocation may occur AFTER the module is unloaded. This is because the module does NOT use stasis_unsubscribe_and_join when terminating the subscription. It merely uses stasis_unsubscribe. This change makes it use stasis_unsubscribe_and_join which is documented for usage in this exact scenario. AST-1520 #close Review: https://reviewboard.asterisk.org/r/4375/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
David M. Lee	702d79de2a	Various fixes for OS X This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sigset() functions. Fixed some format strings for portability. * Fixed compilation issues with res_timing_kqueue (although tests still fail on OS X). * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by: George Joseph ASTERISK-24544 #close Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/4327/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
David M. Lee	197265438e	Add depend on pjproject to res_pjsip_config_wizard.c git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	630eea087d	Investigate and fix memory leaks in Asterisk Fixed memory leaks that were found in Asterisk. ASTERISK-24693 #close Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/4347/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	e23f07beb8	Fix typo's (retrieve, specified, address). ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c73b4b2a46	res_pjsip_outbound_registration.c: Minor code cleanup. * Add an allocation failure check and assert in sip_outbound_registration_response_cb(). * Made sip_outbound_registration_state_destroy() handle partially created state objects from sip_outbound_registration_state_alloc(). Review: https://reviewboard.asterisk.org/r/4366/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	bdfdb01bcf	stasis transfer: fix a race condition on stasis bridge push After a bridge transfer completes where a local replacement channel is used, a stasis transfer message with the details of the transfer is sent. This is processed by stasis which then sets the stasis app name and replaced channel snapshot on the replacement channel. However, since a separate thread was already started to run stasis on the new replacement channel, a race was on to see if the message processing would be completed before the app name was needed, otherwise the channel would be hung up. This change moves the calls used to set the stasis app name and the replace snapshot to the bridge_stasis_push function callback from the bridge transfer logic, allowing the steps to be completed earlier and more deterministically, and the race elimianted. NOTE: the swap channel parameter to bridge_stasis_push (and thus all bridge push callbacks) must always be present when performing a swap with another channel. ASTERISK-24649 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4341/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5e10007dbd	res_pjsip_outbound_registration.c: Move unref to a better place. Move an unconditional unref of client_state so it doesn't look like it could be used after the last ref has destroyed it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Ashley Sanders	a7ba8a58a8	ARI: Fixed crash that occurred when updating a bridge when the optional query parameter 'name' was not supplied. Prior to this changeset, posting to the: /ari/bridges/{bridgeId} endpoint without specifying a value for the [name] query parameter, would crash Asterisk if the bridge you are attempting to create (or update) had the same ID as an existing bridge. The internal mechanism of the POST operation interpreted a null value for name, thus resulting in an error condition that crashed Asterisk. ASTERISK-24560 #close Reported By: Kinsey Moore Review: https://reviewboard.asterisk.org/r/4349/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6af6a216a1	CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge. Calling ast_channel_bridge_peer() cannot be done while holding any channel locks. The reported issue hit the deadlock in chan_iax2, but an audit of the ast_channel_bridge_peer() calls found three more locations where the same deadlock can occur. * Made CHANNEL(peer), res_fax, and the SNMP agent not call ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I had to rework the logic to not hold the channel lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done for legacy reasons that no longer apply. * Removed the iax.conf forcejitterbuffer option. It is now always enabled when the jitterbuffer option is enabled. If you put a jitter buffer on a channel it will be on the channel. ASTERISK-24600 #close Reported by: Jeff Collell Review: https://reviewboard.asterisk.org/r/4342/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	643b81d98e	res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions. The first thing this patch fixes is UAS dialogs. Previously if a transport was configured on an endpoint and an inbound session was created there was no guarantee that requests sent on the dialog would use the correct transport and address information. This has now been fixed so an explicitly configured transport is taken into account. The second thing this patch fixes is res_pjsip_multihomed. The res_pjsip_multihomed module attempts to determine what transport a message should go out on and what addressing information should go into the message itself. In a scenario where multiple transports exist bound to the same IP address but a different port the code would incorrectly alter the transport and change the message to the wrong transport. This change makes the res_pjsip_multihomed module smarter so it will only change the transport and address information in the message when it is possible and makes sense. ASTERISK-24615 #close Reported by: David Justl Review: https://reviewboard.asterisk.org/r/4331/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	34c220203f	REVERTING res_pjsip: make it unloadable Due to the original patch causing memory corruptions the patch is being removed until the problem can be resolved. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	821c15ae53	Fix problem where a hung channel could occur on a failed blind transfer. Different clients react differently to being told that a blind transfer has failed. Some will simply send a BYE and be done with it. Others will attempt to reinvite themselves back onto the call. In the latter case, we were creating a new channel and then leaving it to sit forever doing nothing. With this code change, that new channel will not be created and the dialog with the transferring channel will be cleaned up properly. ASTERISK-24624 #close Reported by Zane Conkle Review: https://reviewboard.asterisk.org/r/4339 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	8bc4a89e1f	Add support for the ca_list_path option for PJSIP transports. This allows for a path to be specified that has a collection of CA certificates in it. ASTERISK-24575 #close Reported by cloos Patches: pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: https://reviewboard.asterisk.org/r/4344 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	fa80d9658d	res_fax.c, res_fax_spandsp.c: Remove redundant locking. When FAX was developed, apparently the faxregistry.container used to be a linked list that was converted to an ao2 container. Some of the replacement ao2 container operations still had explicit lock/unlocks around them. Three off nominal code paths in res_fax.c and res_fax_spandsp.c unlock the channel even though the routine did not lock the channel and other code paths in the routine do not unlock the channel. Review: https://reviewboard.asterisk.org/r/4340/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6c426e86bd	res_fax.c, res_fax_spandsp.c: Fix some curlies on the end of function definitions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	c95391f23c	res_pjsip_outbound_registration: Fix race condition when reloading and listing registrations. Due to the split of outbound registration state from configuration it is possible during a reload for a "pjsip show registrations" CLI command to be executed which gets an older snapshot of the configuration. This configuration may include outbound registrations which have been removed due to a reload operation occurring at the same time. The code for printing the outbound registration did not take this into account but now it does. AST-1506 #close Review: https://reviewboard.asterisk.org/r/4338/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	77a036bf3f	res_pjsip: make it unloadable The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4311/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	e370c9e68e	Prevent slow graceful shutdown when outbound publications never started. The code was missing the case for explicitly destroying an outbound publication when Asterisk had never actually published anything. The result was that Asterisk would hang for a while on a graceful shutdown. With this change, the case is taken into account, and on a graceful shutdown, these publications are destroyed without the need to actually send a PUBLISH request. ASTERISK-24655 #close Reported by Kevin Harwell Review: https://reviewboard.asterisk.org/r/4325 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4b363688d4	AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	eb9ce791d8	res_fax: Add T.38 negotiation timeout option This change makes the T.38 negotiation timeout configurable via 't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously hard coded to be 5000 milliseconds. This change also handles T.38 switch failures by aborting the fax since in the case where this can happen, both sides have agreed to switch to T.38 and Asterisk is unable to do so. Review: https://reviewboard.asterisk.org/r/4320/ ........ Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b937438c17	res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown If you do a 'core (shutdown\|restart) graceful' persistent subscriptions won't survive. If you do a 'core (shutdown\|restart) now' or asterisk terminates for some reason, they do. Here's why... When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to subscribers for each subscription. This not only tells the subscribers that the dialog/state machine is done, it also frees the last reference to the subscription tree which causes the persistent subscription to get deleted from astdb. When asterisk restarts, nothing's left. Just preventing the delete from astdb doesn't work because we already told the subscriber to terminate the dialog so we can't restart it even if it was still in astdb. Everything works OK if asterisk terminates unexpectedly because we never send the 'terminated' message so on restart, the subscription is still in astdb and the subscriber is none the wiser. This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for persistent connections. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4318/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	143bec54ee	res_pjsip_outbound_registration: Fix reference leak. Every time a registration started, sip_outbound_registration_response_cb bumps the ref count on client_state then pushes a handle_registration_response task. handle_registration_response never unreffed it though. So every time a registration goes out, the ref count goes up by one. This patch adds the unreffs to handle_registration_response. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4303/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	6e59bf6491	res_pjsip_outbound_registration: Fix several reload issues There are 2 issues with reloading registrations... 1. The 'can_reuse_registration' test wasn't considering the intervals or expiration in its determination of whether a registration changed or not so if you changed any of the intervals or the expiration and reloaded, the object would get reloaded but the actual timers wouldn't change. can_reuse_registration now does a sorcery diff on the old and new objects instead of discretely testing certain fields. Now if you change expiration for instance, and reload, the timer is updated and re-registration will occur on the new value. 2. If you mung up your password on an outbound registration you get a permanent failure. If you fix the password (on the outbound_auth object) and reload, nothing tells outbound_registration to try again because the registration itself didn't change. This patch adds an observer on the "auth" object type and if any auth changes, existing registration states are searched and those in a REJECTED_PERMANENT state are retried. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4304/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	8f3c60cee7	ARI: Allow usage of ASYNCGOTO with Stasis() When the AMI Redirect action is used with a channel bridged inside Stasis() and not running a pbx, the channel is hung up instead of proceeding to the desired location in dialplan. This change allows such channels to be Redirected properly by detecting the operation used by Redirect (ASYNCGOTO) and using the code already established for functionality of the ARI channel continue operation. ASTERISK-24591 #close Review: https://reviewboard.asterisk.org/r/4271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	42b342c6e2	Add the ability to continue and originate using priority labels. With this patch, the following two ARI commands POST /channels POST /channels/{id}/continue Accept a new parameter, label, that can be used to continue to or originate to a priority label in the dialplan. Because this is adding a new parameter to ARI commands, the API version of ARI has been bumped from 1.6.0 to 1.7.0. This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks! ASTERISK-24412 #close Reported by Nir Simionovich Review: https://reviewboard.asterisk.org/r/4285 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a10d2966b6	res_pjsip_exten_state: Change 'does not exist' warning to notice The 'new_subscribe: Extension <> does not exist or has no associated hint' is a config issue and doesn't need to clutter up logs with warnings. Changed to notice. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4307/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	13ed8f73ed	res_pjsip_mwi: Change "MWI Subscription failed" message from warning to notice The "MWI Subscription failed" message means the client is trying to subscribe to a mailbox that doesn't exist. There's no need to clutter up logs with warnings for a client misconfiguration so I changed it to a notice. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4306/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	9ea8dd036f	Fix ability to perform a remote attended transfer with PJSIP. This fix has two parts: * Corrected an error message to properly state that external_replaces is an extension. The error message also prints what dialplan context the external_replaces extension was being looked for in. * Corrected the printing of the Replaces: header in an INVITE request. We were duplicating "Replaces: " in the header. ASTERISK-24376 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4296 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	e17a1a8ba1	Fix dev-mode build on recent gcc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	4becfae3b1	res_pjsip_mwi: Change warning to notice When res_pjsip loads and an endpoint auto-subscribes a mailbox for mwi, if a contact hasn't registered yet, res_pjsip_mwi spits out a warning. This is a perfectly normal situation though and doesn't require something as serious as a warning. It's also self correcting. The device will start getting mwi as soon as it registers. This patch changes the warning to a notice. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4314/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	0fa6c34dc6	outbound_registration: Add 'pjsip send register' and update 'send unregister' The current behavior of 'pjsip send unregister' is to send the unregister (REGISTER with 0 exp) but let the next scheduled register proceed normally. I don't think that's a good idea. If you unregister, it should stay unregistered until you decide to start registrations again. So this patch just adds a cancel_registration call to the current unregister_task to cancel the timer. Of course, now you need a way to start registration again so I've added a 'pjsip send register' command that unregisters and cancels any existing registration (the same as send unregister), then sends an immediate registration and starts the timer back up again. Both changes also ripple to AMI. There's a new PJSIPRegister command. There's no harm in calling either command repeatedly. They don't care about the actual state. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4301/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d873b09075	pjsip cli: Fix sorting of contacts for 'pjsip list contacts' For some reason I was using a hash container instead of a list to gather the contacts for 'pjsip list/show contacts' so even though I had a sort function, the output wasn't sorted. This patch just changes the hash container to a list container and the contacts now appear sorted in the CLI. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4305/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a7c38428af	pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions. The PJSIP_AOR dialplan function allows inspection of configured AORs including what contacts are currently bound to them. The PJSIP_CONTACT dialplan function allows inspection of contacts in existence. These can include both externally added (by way of registration) or permanent ones. ASTERISK-24341 Reported by: xrobau Review: https://reviewboard.asterisk.org/r/4308/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	cca262e7d3	PJSIP: Update transport method documentation This updates the documentation for the 'method' configuration option to be more verbose about the behaviors of values 'unspecified' and 'default'. They do exactly the same thing which is to select the default as defined by PJSIP which is currently TLSv1. Review: https://reviewboard.asterisk.org/r/4264/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b521c612fc	res_pjsip: Backport missing commits for user_eq_phone This backports the following from trunk, which were missed: r427257 \| file \| 2014-11-04 16:31:16 -0600 (Tue, 04 Nov 2014) \| 2 lines res_pjsip: Allow + at the beginning of a phone number when user_eq_phone is enabled. r427259 \| file \| 2014-11-04 16:51:32 -0600 (Tue, 04 Nov 2014) \| 2 lines res_pjsip: Apply the 'user_eq_phone' setting to the To header as well. It also adds the Alembic script for the option. ASTERISK-24643 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	915bb88d3e	res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports. Note that this is backport from trunk of r425825. This change adds a module which is configurable using the keep_alive_interval setting in the global section that will send a CRLF keep alive to all active connection-oriented transports at the provided interval. This is useful because it can help keep connections open through NATs. This functionality also exists within PJSIP but can not be controlled at runtime and requires recompiling it. Review: https://reviewboard.asterisk.org/r/4084/ ASTERISK-24644 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	006ffdcfb2	res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable. Note that this is a backport of r425804 from trunk. This change adds a configuration option which adds a 'user=phone' parameter if the user portion of the request URI or the From URI is determined to be a number. Review: https://reviewboard.asterisk.org/r/4073/ ASTERISK-24643 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d1c532034b	pjsip_options: Fix continued qualifies after endpoint/aor deletion If you remove an endpoint/aor from pjsip.conf then do a core reload, qualifies will continue even though the object are gone. This happens because nothing clears out the qualify tasks. This patch unschedules all existing qualify tasks before scheduling new ones on reload. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4290/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	fc79cf6428	res_pjsip_phoneprovi_provider: Fix reload Reloading wasn't working correctly because on a reload, the sorcery apply handler was never being called for unchanged users. So, instead of using an apply handler, I'm now iterating over all users. Works much more reliably. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4b054bdc6d	res_http_websocket.c: Fix incorrect use of sizeof in ast_websocket_write(). This won't fix the reported issue but it is an incorrect use of sizeof. ASTERISK-24566 Reported by: Badalian Vyacheslav ........ Merged revisions 429867 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b22c833c12	chan_dahdi.c, res_rtp_asterisk.c: Change some spammy debug messages to level 5. ASTERISK-24337 #close Reported by: Rusty Newton ........ Merged revisions 429804 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	4fad85f9bf	res_pjsip_sdp_rtp: wrong bridge chosen when the DTMF mode is not compatible A native rtp bridge was being chosen (it shouldn't have been) when using two pjsip channels with incompatible DTMF modes. This patch sets the rtp instance property, AST_RTP_PROPERTY_DTMF, for the appropriate DTMF mode(s) for pjsip. It was not being set before, meaning all DTMF modes for pjsip were being treated as compatible, thus native bridging would be chosen as the bridge type when it shouldn't have been. ASTERISK-24459 #close Reported by: Yaniv Simhi Review: https://reviewboard.asterisk.org/r/4265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	14d2f8f20f	Prevent potential infinite outbound authentication loops in registration. Prior to this patch, Asterisk would always respond to 401 responses to registration attempts by trying to provide a registration with authentication credentials. Even if subsequent attempts were rejected with 401 responses, Asterisk would continue this behavior. If authentication credentials were incorrect, this could continue forever. With this patch, we keep track of whether we have attempted authentication on an outbound registration attempt. If we already have, we don not try again until the next attempt. This prevents the infinite loop scenario. Review: https://reviewboard.asterisk.org/r/4273 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b4621cd0f5	res_pjsip_config_wizard: fix unload SEGV If certain pjsip modules aren't loaded, the wizard causes a SEGV when it unloads. Added a check for the presense of the object type wizard before trying to clean it up. Tested-by: George Joseph git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	105f224cfd	res_pjsip_config_wizard: Change FILEUNCHANGED config_load2 flag determination The module now applies the FILEUNCHANGED flag when both reloaded is specified AND there's no last_config for the object type. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4276/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9ae57e0dd6	Fix printf problems with high ascii characters after r413586 (1.8). In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. Those fixes included things like: -out += sprintf(out, "%%%02X", (unsigned char) ptr); +out += sprintf(out, "%%%02X", (unsigned) ptr); That works for low ascii characters, but for the high range that yields e.g. FFFFFFC3 when C3 is expected. This changeset: - fixes those casts to use the 'hh' unsigned char modifier instead - consistently uses %02x instead of %2.2x (or other non-standard usage) - adds a few 'h' modifiers in various places - fixes a 'replcaes' typo - dev/urandon typo (in 13+ patch) Review: https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close Reported by: Stefan27 (on IRC) ........ Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429674 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a3534b7c05	res_pjsip_config_wizard: fix test breakage Fix test breakage caused by not checking for res_pjsip before calling ast_sip_get_sorcery. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4269/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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