55 Commits (523c95e1463553e8d9817a6ea2c2024384247149)

Author	SHA1	Message	Date
Mark Michelson	e24661fd18	Merge Call completion support into trunk. ... From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	42577406fd	Improve handling of T.38 re-INVITEs that arrive before a T.38-capable ... application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	68d1ded8dd	Only change the RTP ssrc when we see that it has changed ... This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	f199367b93	Tweak formatting and add minor updates to some comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a9932363a9	-----Changes ----- ... New files - channels/sip/sip.h – A new header for shared #define, enum, and struct definitions. - channels/sip/include/sip_utils.h – sip util functions shared among the all the sip APIs - channels/sip/include/config_parser.h – sip config-parser API - channels/sip/config_parser.c – Contains sip.conf parsing helper functions with unit tests. - channels/sip/include/reqresp_parser.h – sip request response parser API - channels/sip/reqresp_parser.c – Contains sip request and response parsing helper functions with unit tests. New Unit Tests - sip_parse_uri_test - sip_parse_host_test - sip_parse_register_line_test Code Refactoring - All reusable #define, enum, and struct definitions were moved out of chan_sip.c into sip.h. During this process formatting changes were made to comments in both sip.h and chan_sip.c in order to better adhere to the coding guidelines. - The beginnings of three new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h using existing chan_sip.c functions. - parse_uri() and get_calleridname() were moved from chan_sip.c to request-parser.c along with unit tests for both functions. - sip_parse_host() and sip_parse_register_line() were moved from chan_sip.c to config-parser.c along with unit tests for both functions. Changes to parse_uri() -removal of the options parameter. It was never used and did not behave correctly. -additional check for [?header] field. When this field was present, the transport type was not being set correctly. ----- Overview ----- This patch is introduced with the hope that unit tests for all our sip parsing functions will be written soon. chan_sip is a huge file, and with the addition of each unit test chan_sip is going to grow larger and harder to maintain. I'm proposing we begin refactoring chan_sip, starting with the parsing functions. With each parsing function we move into a separate helper file, a unit test should accompany it. I've attempted to lay down the ground work for this change by creating two new parser helper files (config-parser.c and reqresp-parser.c) and moving all shared structs, enums, and defines from chan_sip.c into a shared sip.h file. We can't verify everything in Asterisk using unit tests, but string parsing is one area where unit tests make the most sense. By beginning to restructure the code in this way, chan_sip not only becomes less bloated, but Asterisk as a whole will become more stable. Review: https://reviewboard.asterisk.org/r/477/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago