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2010-06-08Make SIP tests compile again.rmudgett1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268857 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Mailbox list would previously grow at each reload, containing duplicates.tilghman1-2/+3
Also, optimize the allocation of mailboxes to avoid additional memory structures. (closes issue #16320) Reported by: Marquis Patches: 20100525__issue16320.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268817 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-07Seems strange (and the code backs up) that if the max and min of a statistic ↵tilghman1-3/+3
is expressed as a double, the last value would not also need to be a double. (closes issue #15807) Reported by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268773 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Generic Advice of Charge.rmudgett1-2/+8
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26do all sip registry parsing before transmit_registerdvossel2-23/+153
This patch breaks up every part of the sip registry string during config parsing and removes all parsing from transmit_register(). Thanks to Nick_Lewis for contributing this patch! (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657) chan_sip.c.patch uploaded by Nick Lewis (license 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657) nicklewispatch.diff uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel Review: https://reviewboard.asterisk.org/r/628/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Add support for direct media ACLstwilson1-0/+2
directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address. In some networks not all phones are fully routed, i.e. not all phones can ping each other. This patch adds a way to restrict directmedia for certain peers between certain networks. (closes issue #16645) Reported by: raarts Patches: directmediapermit.patch uploaded by raarts (license 937) Tested by: raarts Review: https://reviewboard.asterisk.org/r/467/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264626 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Enhancements to connected line and redirecting work.mmichelson1-0/+2
From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Permit more lines within a SIP body to be parsed.tilghman1-1/+1
The example given within the related issue showed 120 lines, which was mostly a result of the body being XML. (closes issue #17179) Reported by: khw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261560 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Prevent unnecessary warnings when getting rtpsource or rtpdest.mmichelson1-2/+4
If a recognized media type was present, but the media type was not enabled for the channel, then a warning would be emitted. For instance, attempting to get CHANNEL(rtpsource,video) on a call with no video would cause a warning message to appear. With this change, the warning will only appear if the stream argument is not recognized as being a media type that can be specified. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261313 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Don't override peer context with domain context.mmichelson1-1/+2
(closes issue #17040) Reported by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347) Tested by: pprindeville Review: https://reviewboard.asterisk.org/r/565/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259957 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Be more explicit about field naming in a test.qwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259587 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Add routines for parsing SIP URIs consistently.mmichelson3-56/+982
From the original issue report opened by Nick Lewis: Many sip headers in many sip methods contain the ABNF structure name-andor-addr = name-addr / addr-spec Examples include the to-header, from-header, contact-header, replyto-header At the moment chan_sip.c makes various different attempts to parse this name-andor-addr structure for each header type and for each sip method with sometimes limited degrees of success. I recommend that this name-andor-addr structure be parsed by a dedicated function and that it be used irrespective of the specific method or header that contains the name-andor-addr structure Nick has also included unit tests for verifying these routines as well, so...heck yeah. (closes issue #16708) Reported by: Nick_Lewis Patches: reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis (license 657 Review: https://reviewboard.asterisk.org/r/549 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256530 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson1-2/+363
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: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Add new rtpsource options to the CHANNEL function.mmichelson1-3/+39
This adds rtpsource options analogous to the rtpdest functions that already exist. In addition, this fixes potential crashes which could result due to trying to read values from nonexistent RTP streams. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254551 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Improve handling of T.38 re-INVITEs that arrive before a T.38-capablekpfleming1-37/+41
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: http://svn.digium.com/svn/asterisk/trunk@254450 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Only change the RTP ssrc when we see that it has changedtwilson1-2/+1
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: http://svn.digium.com/svn/asterisk/trunk@252089 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-17Make all of the various rtpqos parameters in this branch available from the ↵tilghman6-3/+567
CHANNEL function. Also includes a test for retrieving rtpqos parameters, including a NULL RTP driver. Additionally, some further separation of the SIP internal API into headers was necessary. (closes issue #16652) Reported by: kkm Patches: 20100204__issue16652.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/501/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247124 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-15chan_sip parse code refactoring plus two new unit testsdvossel3-0/+291
Code Refactoring Changes - read_to_parts() moved to reqresp_parser.c and has been renamed as get_name_and_number() - get_in_brackets() moved to reqresp_parser.c - find_closing_quotes() added to sip_utils.h Logic Changes - get_name_and_number() now uses parse_uri() and get_calleridname() for parsing. Before this change only names within quotes were found, when names not within quotes are possible. New Unit Tests -sip_get_name_and_number_test -sip_get_in_brackets_test (closes issue #16707) Reported by: Nick_Lewis Patches: issue16706.diff uploaded by dvossel (license 671) Review: https://reviewboard.asterisk.org/r/499/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246627 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-11fixes some test description formatting inconsistencies so log file looks nicedvossel2-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246338 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-10additional parse_uri test and documentationdvossel2-1/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246249 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-09Various updates to the unit test API.russell2-87/+73
1) It occurred to me that the difference in usage between the error ast_str and the ast_test_update_status() usage has turned out to be a bit ambiguous in practice. In a lot of cases, the same message was being sent to both. In other cases, it was only sent to one or the other. My opinion now is that in every case, I think it makes sense to do both; we should output it to the CLI as well as save it off for logging purposes. This change results in most of the changes in this diff, since it required changes to all existing unit tests. It also allowed for some simplifications of unit test API implementation code. 2) Update ast_test_status_update() to include the file, function, and line number for the code providing the update. 3) There are some formatting tweaks here and there. Hopefully they aren't too distracting for code review purposes. Reviewboard's diff viewer seems to do a pretty good job of pointing out when something is a whitespace change. 4) I moved the md5_test and sha1_test into the test_utils module. It seemed like a better approach since these tests are so tiny. 5) I changed the number of nodes used in heap_test_2 from 1 million to 100 thousand. The only reason for this was to reduce the time it took for this test to run. 6) Remove an unused function prototype that was at the bottom of utils.h. 7) Simplify test_insert() using the LIST_INSERT_SORTALPHA() macro. The one minor difference in behavior is that it no longer checks for a test registered with the same name. 8) Expand the code in test_alloc() to provide specific error messages for each failure case, to clearly inform developers if they forget to set the name, summary, description, etc. 9) Tweak the output of the "test show registered" CLI command. I swapped the name and category to have the category first. It seemed more natural since that is the sort key. 10) Don't output the status ast_str in the "test show results" CLI command. This is going to tend to be pretty verbose, so just leave that for the detailed test logs (test generate results). Review: https://reviewboard.asterisk.org/r/493/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245864 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-07Tweak formatting and add minor updates to some comments.russell4-21/+40
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245269 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-05fixes issue with sip registry not having correct default expirydvossel2-32/+31
default expiry was not being set correctly for a registry object. Thanks to ebroad for reporting the issue and testing the patch. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244924 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-03-----Changes -----dvossel6-0/+2476
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: http://svn.digium.com/svn/asterisk/trunk@244597 f38db490-d61c-443f-a65b-d21fe96a405b