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2013-09-16app_speech_utils: Fix unresolved symbol ast_speech_get_setting().Richard Mudgett
Fixes regression introduced by -r374096. * Made res_speech.export.in export ast_* symbols instead of specific functions. * Made app_speech_utils.c declare that it is dependent upon res_speech. (issue ASTERISK-17136) Reported by: Richard Kenner ........ Merged revisions 399197 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Restore Dial, Queue, and FollowMe 'I' option support.Richard Mudgett
The Dial, Queue, and FollowMe applications need to inhibit the bridging initial connected line exchange in order to support the 'I' option. * Replaced the pass_reference flag on ast_bridge_join() with a flags parameter to pass other flags defined by enum ast_bridge_join_flags. * Replaced the independent flag on ast_bridge_impart() with a flags parameter to pass other flags defined by enum ast_bridge_impart_flags. * Since the Dial, Queue, and FollowMe applications are now the only callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the calling contract to require the initial COLP exchange to already have been done by the caller. * Made all callers of ast_bridge_impart() check the return value. It is important. As a precaution, I also made the compiler complain now if it is not checked. * Did some cleanup in parking_tests.c as a result of checking the ast_bridge_impart() return value. An independent, but associated change is: * Reduce stack usage in ast_indicate_data() and add a dropping redundant connected line verbose message. (closes issue ASTERISK-22072) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/ ........ Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Create more accurate Contact headers for dialogs when we are the UAS.Mark Michelson
(closes issue AST-1207) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2842 ........ Merged revisions 399083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Change how realms are handled for outbound authentication.Mark Michelson
With this change, if no realm is specified in an outbound auth section, then we will simply match the realm that was present in the 401/407 challenge. (closes issue ASTERISK-22471) Reported by George Joseph (closes issue ASTERISK-22386) Reported by Rusty Newton Patches: outbound_auth_realm_v4.patch uploaded by George Joseph (License #6322) ........ Merged revisions 399059 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Broke the build! Forgot para tags within my description.Rusty Newton
https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304 ........ Merged revisions 399064 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13res_pjsip: Forward PJSIP logging to Asterisk loggingDavid M. Lee
This patch uses PJSIP's pj_log_set_log_func() to forward PJSIP's log messages to Asterisk's logger. This is done in a new module: res_pjsip_log_forwarder.so. This patch sets defaultenabled on the existing res_pjsip_logger.so to no, since logging every SIP packet seems a bit odd to do by default, and is (hopefully) less necessary with regular PJSIP logging. It also removes res_rtp_asterisk's disabling of PJSIP logging. (closes issue ASTERISK-22360) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2830/ ........ Merged revisions 399049 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13ARI: Fix WebSocket response when subprotocol isn't specifiedDavid M. Lee
When I moved the ARI WebSocket from /ws to /ari/events, I added code to allow a WebSocket to connect without specifying the subprotocol if there's only one subprotocol handler registered for the WebSocket. Naively, I coded it to always respond with the subprotocol in use. Unfortunately, according to RFC 6455, if the server's response includes a subprotocol header field that "indicates the use of a subprotocol that was not present in the client's handshake [...], the client MUST _Fail the WebSocket Connection_.", emphasis theirs. This patch correctly omits the Sec-WebSocket-Protocol if one is not specified by the client. (closes issue ASTERISK-22441) Review: https://reviewboard.asterisk.org/r/2828/ ........ Merged revisions 399039 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13'identify' configObject doesn't have a synopsisRusty Newton
Add a straightforward synopsis and description to the identify config object in XML documentation. (issue ASTERISK-22311) (closes issue ASTERISK-22311) Reported By: Rusty Newton ........ Merged revisions 399031 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12Documentation fix and improvements to XML configuration help res_pjsip_aclRusty Newton
* One bug fix. Made the synopsis for "type" to accurate. * changing the usage of "IP-domains" to "IP addresses" * clarifying the usage for the options, by adding a relevant description for each * modified other areas of the XML help for clarity, such as the module description and a few synopsis changes here and there. See the patch. (issue ASTERISK-22458) (closes issue ASTERISK-22458) Reported By: Rusty Newton Review: https://reviewboard.asterisk.org/r/2823/ ........ Merged revisions 399017 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12Fix symbol collision with pjsua.David M. Lee
We shouldn't be exporting any symbols that start with pjsip_. ........ Merged revisions 398927 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11pjsip: reinvite for connected line updates occurs when it should notKevin Harwell
Connected line updates are now only sent out if an actual update needs to occur. This happens under the following conditions: 1. The endpoint we are sending to is trusted. 2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent. 3. The connected id's number and name are valid. Also added an SDP when an update is sent out. (closes issue AST-1212) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2831/ ........ Merged revisions 398806 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-10Fix incorrect usages of ast_realloc().Richard Mudgett
There are several locations in the code base where this is done: buf = ast_realloc(buf, new_size); This is going to leak the original buf contents if the realloc fails. Review: https://reviewboard.asterisk.org/r/2832/ ........ Merged revisions 398757 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398758 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398759 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09Add extra debugging to res_pjsip_endpoint_identifier_ipMark Michelson
........ Merged revisions 398694 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09Multiple revisions 398638-398639David M. Lee
........ r398638 | dlee | 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line Added note about expected behavior of originate ........ r398639 | dlee | 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line Added note about expected behavior of originate (the rest of the commit) ........ Merged revisions 398638-398639 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-07Prevent XMPP timeout on blank responsesKinsey Moore
Sometimes the Google Voice servers have a bad habit of sending out 1 byte replies to the xmpp resource. When a blank 1 byte reply is received from the socket the buffer attempts to wait (endlessly) for the rest of the reply from google which effectively blocks the socket and google voice calls will no longer come into the server. This patch allows the xmpp module to correctly detect empty packets and send out ping replies to google. It also sets a socket timeout on the default socket which prevents the xmpp socket from closing and preventing future google voice calls from coming into the server. Furthermore instead of sending an empty reply back to google we send a proper xmpp ping reply back. This also adds several more socket messages. (closes issue ASTERISK-22347) Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/2771 Patches: xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524) ........ Merged revisions 398618 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398619 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06Multiple revisions 398558,398577Kinsey Moore
........ r398558 | kmoore | 2013-09-06 14:28:16 -0500 (Fri, 06 Sep 2013) | 17 lines Fix Jabber/XMPP distributed MWI The mailbox and context are swapped on the receiving end for all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent versions. This swaps those values to be correct when publishing to the internal event system from Jabber/XMPP distributed MWI state. (closes issue ASTERISK-22435) Reported by: abelbeck Tested by: Michael Keuter Patches: asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck ........ Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) | 10 lines Commit the remainder of r398523 This is a missing part of the commit in revision 398523 that corrects the name of a variable. (issue ASTERISK-22435) ........ Merged revisions 398576 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398558,398577 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398580 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06Fix build warningsKinsey Moore
When AST_DEVMODE is not defined, ast_asserts are not compiled into the binary. In some cases, this means variables are not referenced or are set but unused which causes warnings to show up. (closes issue ASTERISK-22446) Reported by: Jason Parker (qwell) ........ Merged revisions 398521 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-05Clarify server_uri and client_uri registration settings.Mark Michelson
Used some of Rusty's suggested language plus also included more SIPesque descriptions of where the URIs are actually used in an outgoing REGISTER. (closes issue ASTERISK-22390) reported by Rusty Newton ........ Merged revisions 398368 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-04Give more detail regarding failures to create request with auth credentials.Mark Michelson
(issue ASTERISK-22386) ........ Merged revisions 398299 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-04Debug messages for pjsip outbound registrationKevin Harwell
Added debug messages indicating that an outbound registration attempt was made and it was successful in pjsip. (closes issue ASTERISK-22388) Reported by: Rusty Newton ........ Merged revisions 398226 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-03Prevent a crash in res_pjsip_dtmf_info.cKinsey Moore
This change makes sure that a content type header exists before checking the contents of the header against known SIP INFO DTMF content types. ........ Merged revisions 398206 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30Add a reloadable option for sorcery type objectsKevin Harwell
Some configuration objects currently won't place nice if reloaded. Specifically, in this case the pjsip transport objects. Now when registering an object in sorcery one may specify that the object is allowed to be reloaded or not. If the object is set to not reload then upon reloading of the configuration the objects of that type will not be reloaded. The initially loaded objects of that type however will remain. While the transport objects will not longer be reloaded it is still possible for a user to configure an endpoint to an invalid transport. A couple of log messages were added to help diagnose this problem if it occurs. (closes issue ASTERISK-22382) Reported by: Rusty Newton (closes issue ASTERISK-22384) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2807/ ........ Merged revisions 398139 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30Fix various memory leaksKevin Harwell
main/config.c - cleanup cache fie includes res/res_security_log.c - unregister logger level channesl/chan_sip.c - cleanup io context and notify_types main/translator.c - cleanup at shutdown main/named_acl.c - cleanup cli commands main/indications.c - ast_get_indication_tone() unref default_tone_zone if used (closes issues ASTERISK-22378) Reported by: Corey Farrell Patches: config_shutdown.patch uploaded by coreyfarrell (license 5909) res_security_log.patch uploaded by coreyfarrell (license 5909) chan_sip-11.patch uploaded by coreyfarrell (license 5909) indications_refleak.patch uploaded by coreyfarrell (license 5909) named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license 5909) translate_shutdown.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 398102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398103 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398116 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30Memory leak fixKevin Harwell
ast_xmldoc_printable returns an allocated block that must be freed by the caller. Fixed manager.c and res_agi.c to stop leaking these results. (closes issue ASTERISK-22395) Reported by: Corey Farrell Patches: manager-leaks-12.patch uploaded by coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 398060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398061 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398062 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30optional_api: Fix linking problems between modules that export global symbolsDavid M. Lee
With the new work in Asterisk 12, there are some uses of the optional_api that are prone to failure. The details are rather involved, and captured on [the wiki][1]. This patch addresses the issue by removing almost all of the magic from the optional API implementation. Instead of relying on weak symbol resolution, a new optional_api.c module was added to Asterisk core. For modules providing an optional API, the pointer to the implementation function is registered with the core. For modules that use an optional API, a pointer to a stub function, along with a optional_ref function pointer are registered with the core. The optional_ref function pointers is set to the implementation function when it's provided, or the stub function when it's now. Since the implementation no longer relies on magic, it is now supported on all platforms. In the spirit of choice, an OPTIONAL_API flag was added, so we can disable the optional_api if needed (maybe it's buggy on some bizarre platform I haven't tested on) The AST_OPTIONAL_API*() macros themselves remained unchanged, so existing code could remain unchanged. But to help with debugging the optional_api, the patch limits the #include of optional API's to just the modules using the API. This also reduces resource waste maintaining optional_ref pointers that aren't used. Other changes made as a part of this patch: * The stubs for http_websocket that wrap system calls set errno to ENOSYS. * res_http_websocket now properly increments module use count. * In loader.c, the while() wrappers around dlclose() were removed. The while(!dlclose()) is actually an anti-pattern, which can lead to infinite loops if the module you're attempting to unload exports a symbol that was directly linked to. * The special handling of nonoptreq on systems without weak symbol support was removed, since we no longer rely on weak symbols for optional_api. [1]: https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue ASTERISK-22296) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2797/ ........ Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30ARI: Implement /recordings/stored API'sDavid M. Lee
his patch implements the ARI API's for stored recordings. While the original task only specified deleting a recording, it was simple enough to implement the GET for all recordings, and for an individual recording. The recording playback operation was modified to use the same code for accessing the recording as the REST API, so that they will behave consistently. There were several problems with the api-docs that were also fixed, bringing the ARI spec in line with the implementation. There were some 'wishful thinking' fields on the stored recording model (duration and timestamp) that were removed, because I ended up not implementing a metadata file to go along with the recording to store such information. The GET /recordings/live operation was removed, since it's not really that useful to get a list of all recordings that are currently going on in the system. (At least, if we did that, we'd probably want to also list all of the current playbacks. Which seems weird.) (closes issue ASTERISK-21582) Review: https://reviewboard.asterisk.org/r/2693/ ........ Merged revisions 397985 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30Sanitize XML output for PIDF bodies.Mark Michelson
PJSIP's PIDF API does not replace angle brackets with their appropriate counterparts for XML. So we have to do it ourself. In this particular case, the problem had to do with attempting to place an unsanitized SIP URI into an XML node. Now we don't get a 488 from recipients of our PIDF NOTIFYs. ........ Merged revisions 397968 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-29Fix method for creating activities string in PIDF bodies.Mark Michelson
The previous method did not allocate enough space to create the entire string, but adjusted the string's slen value to be larger than the actual allocation. This resulted in garbled text in NOTIFY requests from Asterisk. This method allocates the proper amount of space first and then writes the content into the buffer. ........ Merged revisions 397960 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-29Fix when the subscription_terminated callback is called for subscription ↵Mark Michelson
handlers. The previous placement would result in the resubscribe() callback called instead of the subscription_terminated() callback being called when a subscription was ended via a SUBSCRIBE request. This would result in confusing PJSIP and having it throw an assertion. ........ Merged revisions 397955 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-29Fix a race condition where a canceled call was answered.Mark Michelson
RFC 5407 section 3.1.2 details a scenario where a UAC sends a CANCEL at the same time that a UAS sends a 200 OK for the INVITE that the UAC is canceling. When this occurs, it is the role of the UAC to immediately send a BYE to terminate the call. This scenario was reproducible by have a Digium phone with two lines place a call to a second phone that forwarded the call to the second line on the original phone. The Digium phone, upon realizing that it was connecting to itself, would attempt to cancel the call. The timing of this happened to trigger the aforementioned race condition about 80% of the time. Asterisk was not doing its job of sending a BYE when receiving a 200 OK on a cancelled INVITE. The result was that the ast_channel structure was destroyed but the underlying SIP session, as well as the PJSIP inv_session and dialog, were still alive. Attempting to perform an action such as a transfer, once in this state, would result in Asterisk crashing. The circumstances are now detected properly and the session is ended as recommended in RFC 5407. (closes issue AST-1209) reported by John Bigelow ........ Merged revisions 397945 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-28Improve detection of answer on SIP blind transfer.Mark Michelson
A problem encountered during testing was that res_pjsip_refer would not ever send a NOTIFY with a 200 OK sipfrag. This is because the framehook that was supposed to send the NOTIFY would never be told that an answer had occurred. This happened for two reasons: 1) The transferee channel on which the framehook was on was already up. 2) Answers are rarely if ever written to channels. Rather, the ast_answer() or ast_raw_answer() function is used to answer channels. Thanks to a suggestion by Matt Jordan, the best way to detect that the call had been answered was to find out when the transferee channel joined a bridge. With stasis this is an easy task. So now, in addition to the framehook logic, there is a stasis subscription used to determine when the transferee has entered a bridge. Once it has entered, an appropriate NOTIFY is sent. ........ Merged revisions 397876 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-28Fix dialog matching in the SIP distributor.Mark Michelson
Dialog matching is performed in the distributor for the sole purpose of retrieving an associated serializer so the request may be serialized. This patch fixes two problems. First, incoming CANCEL requests that had no to-tag (which really should be *all* CANCEL requests) would not match with a dialog. An earlier bug fix to deal with early CANCEL requests would result in the CANCEL being replied to with a 481. The fix for this is to find the matching INVITE transaction and get the dialog from that transaction. Second, no SIP responses were matching dialogs. This is because we were inverting the tags that we were passing into PJSIP's dialog finding function. This logic has been corrected by setting local and remote tag variables based on whether the incoming message is a request or response. ........ Merged revisions 397854 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-27ARI: WebSocket event cleanupDavid M. Lee
Stasis events (which get distributed over the ARI WebSocket) are created by subscribing to the channel_all_cached and bridge_all_cached topics, filtering out events for channels/bridges currently subscribed to. There are two issues with that. First was a race condition, where messages in-flight to the master subscribe-to-all-things topic would get sent out, even though the events happened before the channel was put into Stasis. Secondly, as the number of channels and bridges grow in the system, the work spent filtering messages becomes excessive. Since r395954, individual channels and bridges have caching topics, and can be subscribed to individually. This patch takes advantage, so that channels and bridges are subscribed to on demand, instead of filtering the global topics. The one case where filtering is still required is handling BridgeMerge messages, which are published directly to the bridge_all topic. Other than the change to how subscriptions work, this patch mostly just moves code around. Most of the work generating JSON objects from messages was moved to .to_json handlers on the message types. The callback functions handling app subscriptions were moved from res_stasis (b/c they were global to the model) to stasis/app.c (b/c they are local to the app now). (closes issue ASTERISK-21969) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2754/ ........ Merged revisions 397816 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-25Add rtpengine configuration parameterMatthew Jordan
The rtpengine configuration parameter was documented in the XML documentation, but it was not actually registered with the sorcery object. This adds the parameter with a default of "asterisk", such that res_rtp_asterisk is chosen as the default RTP implementation. (closes issue ASTERISK-22380) Reported by: Rusty Newton Tested by: Rusty Newton ........ Merged revisions 397621 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Add some clarifying documentation to the rewrite_contact endpoint option.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Handle DTMF and hold wrapup when a channel leaves the bridging system.Richard Mudgett
DTMF start/end and hold/unhold events have state because a DTMF begin event and hold event must be ended by something. The following cases need to be handled when a channel is moved around in the system. * When a channel leaves a bridge it may owe a DTMF end event to the bridge. * When a channel leaves a bridge it may owe an UNHOLD event to the bridge. (This case is explicitly ignored because things like transfers need explicit control over this.) * When a channel leaves the bridging system it may need to simulate a DTMF end event to the channel. * When a channel leaves the bridging system it may need to simulate an UNHOLD event to the channel. The patch also fixes the following: * Fixes playing a file and restarting MOH using the latest MOH class used. (closes issue ASTERISK-22043) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2791/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23ARI: Correct error codes for bridge operationsDavid M. Lee
This patch adds error checking to ARI bridge operations, when adding/removing channels to/from bridges. In general, the error codes fall out as follows: * Bridge not found - 404 Not Found * Bridge not in Stasis - 409 Conflict * Channel not found - 400 Bad Request * Channel not in Stasis - 422 Unprocessable Entity * Channel not in this bridge (on remove) - 422 Unprocessable Entity (closes issue ASTERISK-22036) Review: https://reviewboard.asterisk.org/r/2769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Add pass through support for Opus and VP8; Opus format attribute negotiationMatthew Jordan
This patch adds pass through support for Opus and VP8. That includes: * Format attribute negotiation for Opus. Note that unlike some other codecs, the draft RFC specifies having spaces delimiting the attributes in addition to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in chan_sip, so a small tweak was also included in this patch for that. * A format attribute negotiation module for Opus, res_format_attr_opus * Fast picture update for VP8. Since VP8 uses a different RTCP packet number than FIR, this really is specific to VP8 at this time. Note that the format attribute negotiation in res_pjsip_sdp_rtp was written by mjordan. The rest of this patch was written completely by Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/ (closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches: asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Update config framework/sorcery with types/options without documentationMatthew Jordan
There are times when a configuration option should not have documentation. 1. Some options are registered with a particular object merely as a warning to users. These options aren't even really 'deprecated' - which has its own separate API call - they are actually provided by a different configuration file. The options are merely registered so that the user gets a warning that a different configuration file provides the item. 2. Some object types - most notably some used by modules that use sorcery - are completely internal and should never be shown to the user. 3. Sorcery itself has several 'hidden' fields that should never be shown to a user. This patch updates the configuration framework and sorcery with additional API calls that allow a module to register types as internal and options as not requiring documentation. This bypasses the XML documentation checking. This patch also re-enables the strict XML documentation checking in trunk, as well as updates some documentation that was missing. Review: https://reviewboard.asterisk.org/r/2785/ (closes issue ASTERISK-22359) Reported by: Matt Jordan (closes issue ASTERISK-22112) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23ARI: Music on Hold/Background Music for bridgesJonathan Rose
Adds ARI functions to be able to turn on/off music on hold in a bridge. It actually functions more as a background music without further actions on the bridge since if the rest of the channels in the bridge aren't explicitly muted, they will still be able to communicate. (closes issue ASTERISK-21974) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Add SayAlphaCase and similar functionality for AGIKinsey Moore
This adds a new dialplan application, SayAlphaCase, that performs much the same function as SayAlpha except that it takes additional options which allow the user to specify whether the case of each letter should be announced for uppercase, lowercase, or all letters. Similar functionality has been added to the SAY ALPHA AGI command via an optional parameter. Original Patch by: Kevin Scott Adams Reported by: Kevin Scott Adams Review: https://reviewboard.asterisk.org/r/2725/ (closes issue ASTERISK-20782) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22res_sip_dtmf_info: Support sending of 'raw' DTMFKevin Harwell
Added the ability to handle 'raw' DTMF within the body of an INFO message. Also made it so values 10-16 are mapped to valid DTMF values. (closes issue ASTERISK-22144) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2776/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Add missing configOption close tagsKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Fix missing xml doc configOption 'type' for for both 'system' and 'global' ↵Rusty Newton
configObjects (issue ASTERISK-22344) (closes issue ASTERISK-22344) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Bridge API: Set a cause code on a channel when it is ejected from a bridge.Richard Mudgett
The cause code needs to be passed from the disconnecting channel to the bridge peers if the disconnecting channel dissolves the bridge. * Made the call to an app_agent_pool agent disconnect with the busy cause code if the agent does not ack the call in time or hangs up before acking the call. (closes issue ASTERISK-22042) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2772/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Handle default body types for SIP event packages in res_pjsip_pubsubMark Michelson
Prior to this change, we would reject SUBSCRIBE requests that had no Accept headers. Now event package handlers that handle the default type for the event package indicate that they do so. Therefore, if we have a handler that can handle the default type, we can allow SUBSCRIBEs for the handler's event package that have no Accept headers. (closes issue ASTERISK-22067) reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/2774 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21ARI: Correct segfault with /variable calls are missing ?variable parameter.David M. Lee
Both /asterisk/variable and /channel/{channelId}/variable requires a ?variable parameter to be passed into the query. But we weren't checking for the parameter being missing, which caused a segfault. All calls now properly return 400 Bad Request errors when the parameter is missing. The Swagger api-docs were updated accordingly. (closes issue ASTERISK-22273) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21res_stasis: remove call to missing function control_continue.David M. Lee
In the shuffling around of res_stasis, control_continue was renamed to stasis_app_control_continue, but the call in res_stasis wasn't updated. In looking into it, it turns out it wasn't really the right thing to do in res_stasis anyways. This patch changes the handling of received a AST_CONTROL_HANGUP frame to be the same as receiving a NULL frame, and removed the declaration of control_continue(), since it doesn't exist any more. (closes issue ASTERISK-22292) Reported by: Denis Smirnov git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Fix several interrelated issues dealing with the holding bridge technology.Richard Mudgett
* Added an option flags parameter to interval hooks. Interval hooks now can specify if the callback will affect the media path or not. * Added an option flags parameter to the bridge action custom callback. The action callback now can specify if the callback will affect the media path or not. * Made the holding bridge technology reexamine the participant idle mode option whenever the entertainment is restarted. * Fixed app_agent_pool waiting agents needlessly starting and stopping MOH every second by specifying the heartbeat interval hook as not affecting the media path. * Fixed app_agent_pool agent alert from restarting the MOH after the alert beep. The agent entertainment is now changed from MOH to silence after the alert beep. * Fixed holding bridge technology to defer starting the entertainment. It was previously a mixture of immediate and deferred. * Fixed holding bridge technology to immediately stop the entertainment. It was previously a mixture of immediate and deferred. If the channel left the bridging system, any deferred stopping was discarded before taking effect. * Miscellaneous holding bridge technology rework coding improvements. Review: https://reviewboard.asterisk.org/r/2761/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Allow channels in app_stasis to hangup properlyKinsey Moore
This detects hangups that occur while bridged to allow channels to exit app_stasis even if the hangup frame was absorbed by the bridge the channel was in. Reported by: David Lee (closes issue ASTERISK-22297) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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