asterisk

Commit Graph

Author	SHA1	Message	Date
Andrew Latham	42a8af5c9a	CREDITS clean up As discussed online http://lists.digium.com/pipermail/asterisk-dev/2012-October/057245.html the credits file needs some cleaning. This is 95% whitespace with a few additions found in file headers. Further additions should be added here instead of in the file being updated. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	5627cd6f6b	Revert Local testing Config Revert a local testing config that I made. This was not intended to be committed. Thank you Matt Jordan for noticing this. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	ccb7b3a1b5	Fix a bug where audio on Google Voice would not work due to ignoring candidates. Instead of ignoring parts of the message that are not known just ignore the ones we know may be present and that would cause a problem. ........ Merged revisions 374877 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	cd9745be1b	Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures. This change removes the requirement for ufrag and pwd in the transport stanza and also makes us the controlling agent. (closes issue ASTERISK-20554) Reported by: mmichelson ........ Merged revisions 374850 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	825607e09b	Don't make chan_sip export global symbols. During testing, it was discovered that having chan_sip export global symbols was problematic. The biggest problem was that load order was affected. Trying to use realtime could be problematic since in all likelihood the necessary realtime driver(s) would not be loaded before chan_sip. In addition, it was found that it was impossible to use the Digium Phone Module for Asterisk since it must be loaded before chan_sip since it must hook into chan_sip's configuration parsing. The solution is to use a virtual table in the same manner that other modules in Asterisk do, like app_voicemail. (closes issue ASTERISK-20545) Reported by: kmoore ........ Merged revisions 374842 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374849 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5ac43a08ad	Fix incorrect billing duration reported when batch mode is enabled Similar to r369351, the billing duration can be skewed when batch mode is enabled. This happened much more rarely than the duration, as it only occured when the call was answered (thereby indicating an actual answer time) and immediately hung up on (indicating a billsec of 0). Since a billing time of '0' can either mean that the call immediately ended or that the CDR was improperly answered, we have to use additional information to know whether or not we can trust the CDR billsec value. Prior to this patch, we looked to see if we had a valid answer time. If we did, and billsec was zero, we used the current time to calculate what billsec value we could from the CDR being written. If batch mode is enabled, this will incorrectly report a billsec value being much greater than the actual duration of the call. Instead of relying on the presence of an answer time to know whether or not we can re-calculate the billsec for the CDR, we now also use the presence of the CDR's end time to know if we need to re-calculate or whether we can trust the billsec value that we have. This prevents erroneous jumps in the billsec value, while still making sure that in the worst case, some billing time will be calculated. (closes issue AST-1016) Reported by: Thomas Arimont Tested by: Thomas Arimont ........ Merged revisions 374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374844 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374845 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	755c2b8708	Consider the Google Talk content stanza name (jin:content) valid. ........ Merged revisions 374833 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	516c9ec665	app_queue: Made pass connected line updates from the caller to ringing queue members. Party A calls Party B Party B puts Party A on hold. Party B calls a queue. Ringing queue member D sees Party B identification. Party B transfers Party A to the queue. Queue member D does not get a connected line update for Party A. Queue member D answers the call and still sees Party B information. However, if Party A later transfers the call to Party C then queue member D gets a connected line update for Party C. * Made pass connected line updates from the caller to queue members while the queue members are ringing. (closes issue AST-1017) Reported by: Thomas Arimont (closes issue ABE-2886) Reported by: Thomas Arimont Tested by: rmudgett ........ Merged revisions 374801 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 374802 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374803 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374804 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	357b733b66	Fix segfault regression from r370681 Due to usage of ast_hook_send_action, AMI action handling code should be able to handle a NULL mansession->session. This would cause a crash on NULL dereference if action_originate was called from ast_hook_send_action. (closes issue ASTERISK-20544) ........ Merged revisions 374792 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9a0ff62452	Fix execution of 'i' extension due to uninitialized variable. The fix for ASTERISK-18243 added code that could potentially use dst_exten[] uninitialized. As a result the 'i' exten may not be executed when it should. (closes issue ASTERISK-20455) Reported by: Richard Miller Patches: pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard Miller Made some cosmetic modifications. ........ Merged revisions 374758 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374763 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374771 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	766d133c62	Improve logging for DTLS-SRTP failure situations. (closes issue ASTERISK-20487) Reported by: mjordan ........ Merged revisions 374756 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3e1d2917bb	dahdi.conf.sample: Add description for "buffers" setting. This contains an edited version of the patch originally created by John Bigelow. (closes issue ASTERISK-14435) Reported by: John Bigelow Patches: buffers.patch (license #5091) patch uploaded by John Bigelow 0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch (license #5417) patch uploaded by Shaun Ruffell Modified ........ Merged revisions 374727 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374728 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374729 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ab548bb0ce	Fix deletion of unopenable spool files. If scan_service() cannot open the spool file, it logs a message saying that it will delete the file and calls remove_from_queue() to do it. However, remove_from_queue() fails to delete the spool file because struct outgoing has not yet been fully initialized. * Merged allocating a new struct outgoing and init_outgoing() into new_outgoing(). Allocation is initialization. * Made apply_outgoing() not initialize the spool filename in struct outgoing. * Made apply_outgoing() call ast_trim_blanks() and ast_skip_blanks() rather than manually inlining them. * Reduced indentation levels in apply_outgoing(). * Fixed a garbled comment in remove_from_queue(). * Reworked scan_service() to simplify it. (closes issue ASTERISK-17231) Reported by: David Chappell Patches: spool_open_failure.diff (license #4997) patch uploaded by David Chappell Started with this patch. ........ Merged revisions 374686 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixed some memory leaks on off nominal paths in init_outgoing() when merging into the new_outgoing() function dealing with o->capabilities. ........ Merged revisions 374695 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374708 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3620fcff36	Disable ICE support by default Since there are a number of legacy devices out there that fail to handle ICE candidates properly (which is a nice way of saying something much uglier), disable it by default. Support for ICE candidates can be enabled in rtp.conf using the icesupport setting. ........ Merged revisions 374676 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	be906d6318	Resolve issues in ConfBridge regarding marked, waitmarked, and unmarked users Thank's to Neil Tallim (flan)'s tireless testing, issue reporting, and patches it became clear that app_confbridge had some complex logic in how it handled interactions between marked, waitmarked, and unmarked users. In particular, there were some areas in which the interactions between the users resulted in inconsistent behavior, and app_confbridge was missing logic in how to handle some corner cases. Some areas included: * Poor handling of mixing unmarked and waitmarked users * Inconsistencies in how MOH and muting was applied to various users * Handling of various announcements for different user profile options flan's patches seem to fix the various issues, but highlighted how hard the code could be to maintain. In an attempt to make things easier to maintain and to more fully enumerate the various cases that exist, this patch breaks up the logic into a state machine-like setup. Please note that the various state transitioned are documented on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes Review: //https://reviewboard.asterisk.org/r/2072/ Note that for the following issues, mjordan uploaded the patch, although it was written by twilson. Any contributor license discrepency is due to that. (closes issue ASTERISK-19562) Reported by: flan Tested by: flan, mjordan, jrose patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) (closes issue ASTERISK-19726) Reported by: flan Tested by: flan patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) (closes issue ASTERISK-20181) Reported by: Jonathan White Tested by: Jonathan White patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) ........ Merged revisions 374652 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374657 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	35b12af8b6	pjproject: Fix for Solaris builds. Do not undef s_addr. pjproject, in order to solve build problems on Windows [1], undefines s_addr in one of it's headers that is included in res_rtp_asterisk.c. On Solaris s_addr is not a structure member, but defined to map to the real strucuture member, therefore when building on Solaris it's possible to get build errors like: [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o In file included from /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29, from res_rtp_asterisk.c:51: /export/home/admin/asterisk-11-svn/include/asterisk/network.h: In function `inaddrcmp': /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr' /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr' res_rtp_asterisk.c: In function `ast_rtp_on_ice_tx_pkt': res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer will break strict-aliasing rules res_rtp_asterisk.c:710: warning: dereferencing type-punned pointer will break strict-aliasing rules res_rtp_asterisk.c: In function `rtp_add_candidates_to_ice': res_rtp_asterisk.c:1085: error: structure has no member named `s_addr' make[2]: * [res_rtp_asterisk.o] Error 1 make[1]: * [res] Error 2 make[1]: Leaving directory `/export/home/admin/asterisk-11-svn' gmake: *** [_cleantest_all] Error 2 Unfortunately, in order to make this work, I also had to make sure pjproject only used the typdef pj_in_addr and not the struct pj_in_addr so that when building Asterisk I could "typedef struct in_addr pj_in_addr". It's possible then that the library and users of those interfaces in Asterisk have a different idea about the type of the argument, while on the surface it looks like they are all 32 bit big endian values. [1] http://trac.pjsip.org/repos/changeset/484 (issues ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang, mjordan patches: 0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch uploaded by Shaun Ruffell (license 5417) ........ Merged revisions 374642 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	6b5677a41d	Trivial patch to make 'best_score' defined for all architectures. Fixes trivial build error on Solaris: acl.c: In function `get_local_address': acl.c:196: error: `best_score' undeclared (first use in this function) acl.c:196: error: (Each undeclared identifier is reported only once acl.c:196: error: for each function it appears in.) make[2]: *** [acl.o] Error 1 (issue ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang patches: 0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch by Shaun Ruffell (license 5417) ........ Merged revisions 374632 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	bd36827e98	Handle capability stanzas that fail to provide node or version information While XEP-0115 states that the node and ver attributes are both required, some devices fail to provide either field. Prior to this patch, failure to provide the node or ver attribute would cause a crash in res_xmpp. While failing to provide the node or ver attribute is technically invalid, since this information is not utilized by Asterisk except for reporting purposes, for interoperability reasons, we continue to process the capability stanza anyways. (closes issue ASTERISK-20495) Reported by: Martin W Tested by: Martin W patches: 20495.patch uploaded by Martin W (license #6434) ........ Merged revisions 374622 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	15b35972ff	Update documentation for MessageSend application/command's From field for XMPP When using the channel technology agnostic application/AMI command MessageSend, the "From" field is technically optional for the SIP channel driver. However, if being sent by the XMPP resource module (either res_xmpp or res_jabber), the "From" field is necessary, and must correspond to a defined account. This patch updates the documentation for this application/AMI command to reflect this. (closes issue ASTERISK-20405) Reported by: Leif Madsen ........ Merged revisions 374611 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	5f9f6f3867	Multiple revisions 374570,374581 ........ r374570 \| dlee \| 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) \| 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 \| dlee \| 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) \| 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374586 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374587 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	79baef5bbd	Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 \| rmudgett \| 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) \| 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 \| rmudgett \| 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) \| 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 \| rmudgett \| 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) \| 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 \| rmudgett \| 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) \| 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 \| rmudgett \| 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) \| 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 \| rmudgett \| 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) \| 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 \| rmudgett \| 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) \| 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 \| rmudgett \| 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) \| 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 \| rmudgett \| 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) \| 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 \| rmudgett \| 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) \| 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 \| rmudgett \| 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) \| 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 \| rmudgett \| 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) \| 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	90f8c90b10	dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of a recompile, allow values to be adjusted in dsp.conf For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2144/ ........ Merged revisions 374479 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374481 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374485 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	36405ae659	dsp.c fix incorrect DTMF Digit_Duration. it's always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2145/ ........ Merged revisions 374475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374476 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374477 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	c5acf22cec	Fix DBDelTree error codes for AMI, CLI and AGI The AMI DBDelTree command will return Success/Key tree deleted successfully even if the given key does not exist. The CLI command 'database deltree' had a similar problem, but was saved because it actually responded with '0 database entries removed'. AGI had a slightly different error, where it would return success if the database was unavailable. This came from confusion about the ast_db_deltree retval, which is -1 in the event of a database error, or number of entries deleted (including 0 for deleting nothing). * Changed some poorly named res variables to num_deleted * Specified specific errors when calling ast_db_deltree (database unavailable vs. entry not found vs. success) * Fixed similar bug in AGI database deltree, where 'Database unavailable' results in successful result (closes issue AST-967) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2138/ ........ Merged revisions 374426 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374427 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374428 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	d78f7f92b2	Add support for applying direct media ACLs between differing channel technologies. Review: https://reviewboard.asterisk.org/r/2122/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	4af961a03a	dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries. Various countries have different specifications for the maximum power level differences between the DTMF low group and high group of frequencies. Power level difference between frequencies for different Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03) Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T specifications Add's the following variables to dsp.conf ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51 ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98 (closes issue ASTERISK-20442) Reported by: tbsky Tested by: tbsky,alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2141/ ........ Merged revisions 374384 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374385 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374386 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	481df22eac	Check for presence of buddy in info/dinfo handlers The res_jabber resource module uses the ASTOBJ library for managing its ref counted objects. After calling ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to the object has to be checked to see if the buddy existed. Prior to this patch, the buddy object was not checked for NULL; with this patch in both aji_client_info_handler and aji_dinfo_handler the pointer is checked before used and, if no buddy object was found, the handlers return an error code. This patch does not take the approach that our JID can be used to log in from another resource. If that approach is desired, an improvement could be made to this patch to create the buddy on the fly. This patch seeks only to prevent Asterisk from crashing. FYI: In Asterisk 11+, you really should be using res_xmpp. It does not have this problem, as it moved to the astobj2 library. Note that multiple people have proposed patches for this issue; the patch being committed here is based on those. (closes issue ASTERISK-19532) Reported by: Karsten Wemheuer Tested by: Byron Clark patches: fix-jabber uploaded by Karsten Wemheuer (license #5930) xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark (license #6157) (closes issue ASTERISK-19557) Reported by: ulugutz ........ Merged revisions 374335 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374336 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374337 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	9367893cff	Destroy the generic_monitors container after the core_instances in ccss For each item in core_instances disposed of in the shutdown of ccss, any generic monitor instances referenced by the objects will be removed from generic_monitors during their destruction. Hilarity ensues if generic_monitors no longer exists. Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly when it ran into this. ........ Merged revisions 374300 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374301 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9b25076e83	Missed an astobj2.c debug tag. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	427abb240a	* Add ref debug tags to astobj2.c ref usage. * Make container nodes not show up in the ref debug log. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	bfe35ee0b0	Ensure Shutdown AMI event is still fired during Asterisk shutdown Richard pointed out that having the manager dispose of itself gracefully during shutdown meant that the Shutdown event will no longer get fired. This patch moves the AMI event just prior to running the atexit callbacks. ........ Merged revisions 374230 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374231 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374248 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0be8d5c64c	Modify hashtest2 to compile after r374213. Someone, somewhere, may care. Because hashtest2 has to provide symbols for things in asterisk that items it includes may use, when astobj2 decided to use ast_register_atexit it needed to provide a declaration for that as well. Otherwise - no linky. On a related note, ASTERISK-20505 was filed to convert hashtest/hashtest2 into actual unit tests, so we don't run into this problem again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ed3acdeb21	Fix findings from check-in on r374177 Richard pointed out two problems with the check-in from r374177: * The ast_msg_shutdown function declaration doesn't match the prototype in main/message.c. * The ref/alloc function usage in astobj2 (in trunk) can use the ao2_t_* variants of the functions to allow the REF_DEBUG flag to enable/disable their debug counterparts. ........ Merged revisions 374210 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374211 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a094707d51	Fix a variety of ref counting issues This patch resolves a number of ref leaks that occur primarily on Asterisk shutdown. It adds a variety of shutdown routines to core portions of Asterisk such that they can reclaim resources allocate duringd initialization. Review: https://reviewboard.asterisk.org/r/2137 ........ Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374178 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374196 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	4e228fce03	Doxygen Cleanup Start adding configuration file linking and pages. Add module loading doxygen block. Breaking up commits to keep it easy to track (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	99e1174bfa	Doxygen Cleanup Start adding configuration file linking and pages. Add module loading doxygen block. Breaking up commits to keep it easy to track (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	e11cc29360	Doxygen Cleanup Start adding configuration file linking and pages. Add module loading doxygen block. Breaking up commits to keep it easy to track (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	14be2a5514	Doxygen Cleanup Start adding configuration file linking and pages. Add module loading doxygen block. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	b9eeff1521	app_queue: Support persisting and loading of long member lists. Greenlight in #asterisk brought up that he was receiving an error message "Could not create persistent member string, out of space" when running app_queue in Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to store the generated string, but with queues that have large member lists this is not always the case. This patch removes the limitation and uses ast_str instead of a fixed sized buffer. The complicating factor comes from the fact that ast_db_get requires a buffer and buffer size argument, which doesn't let us pull back more than what we pass in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d copy of the value from astdb. As an aside, I did some testing on the maximum size of data that we can store in the BDB library we distribute and was able to store a 10MB string and retrieve it with no problems, so I feel this is a safe patch. Review: https://reviewboard.asterisk.org/r/2136/ ........ Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374135 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374150 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	b3c739a842	Use ast_copy_string instead of strncpy to guarantee a NUL terminated string. ........ Merged revisions 374132 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374133 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	154ee24ae8	Change core show help output format. The CLI "core show help" output leaves something to be desired. 1) The command is truncated to a maximum of 30 characters. 2) The output columns are mirrored from the 31st column. Current output format: logger mute Toggle logging output to a console logger reload Reopens the log files logger rotate Rotates and reopens the log files logger set level {DEBUG\|NOTICE Enables/Disables a specific logging level for this console logger show channels List configured log channels New format: logger mute -- Toggle logging output to a console logger reload -- Reopens the log files logger rotate -- Rotates and reopens the log files logger set level {DEBUG\|NOTICE\|WARNING\|ERROR\|VERBOSE\|DTMF} {on\|off} -- Enables/Disables a specific logging level for this console logger show channels -- List configured log channels Review: https://reviewboard.asterisk.org/r/2133/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	d819508bae	Don't destroy confbridge config when error is encountered during a reload. Not panicking means that the old config is kept. (closes issue ASTERISK-20458) Reported by: Leif Madsen Patches: ASTERISK-20458.patch uploaded by Mark Michelson(license #5049) Tested by Leif Madsen ........ Merged revisions 374106 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	0fc114dc65	Add support for retrieving engine specific settings using the speech API and from dialplan. (closes issue ASTERISK-17136) Reported by: kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c3c317433f	Fix ref leak when adding ICE candidates to an SDP There was a missing decrement to the reference count for the current ICE candidate when local candidates are being added to an outbound SDP. This patch corrects that. ........ Merged revisions 374085 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1f9d7090df	Include channel uniqueid in "AsyncAGI" and "AGIExec" events. * Added AMI event documentation for AsyncAGI and AGIExec events. (closes issue ASTERISK-20318) Reported by: Dan Cropp Patches: res_agi_patch.txt (license #6422) patch uploaded by Dan Cropp modified for trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	02d2280543	res_jabber: Remove CLI command 'jabber test' The opinion of development was that it is both improper to have Matt's personal email address used in the source and that the command wouldn't be useful without it. (closes issue AST-467) Reported by: Malcolm Davenport ........ Merged revisions 374032 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374045 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374059 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b5138fccf4	Add pause one second W dial modifier. * The following dialplan applications now recognize 'W' to pause sending DTMF for one second in addition to the previously existing 'w' that paused sending DTMF for half a second. Dial, ExternalIVR, and SendDTMF. * The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now distinguishes between 'w' and 'W'. The 'w' pauses dialing for half a second. The 'W' pauses dialing for one second. * Created dahdi_dial_str() in chan_dahdi that eliminated a lot of duplicated dialing code and diagnostic messages for the channel driver. (closes issue ASTERISK-20039) Reported by: Jeremiah Gowdy Patches: jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy Expanded patch to add support in chan_dahdi. Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Brent Eagles	89d427ca24	Reset hangup flags on channels created through messages and cleanup globals in res_xmpp on unload. This patch fixes an issue where hangup flags were not being reset on a channel, affecting subsequent use of that channel. The patch also adds some additional cleanup to res_xmpp to fix an issue with reloading the module. (closes ASTERISK-20360) Reported by: Noah Engelberth Tested by: beagles Review: https://reviewboard.asterisk.org/r/2134/ ........ Merged revisions 374019 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	b6a00a1d97	Update documentation to make it explicit that "stream file" will not restart musiconhold. (issue ASTERISK-17367) Reported by: oej ........ Merged revisions 373989 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373990 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373991 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	acb3a5f76f	Add Duration header for PlayDTMF AMI Action This patch adds an optional header to the PlayDTMF AMI action, Duration. It allows the duration of the DTMF digit to be played on the channel to be specified in milliseconds. (closes issue ASTERISK-18172) Reported by: Renato dos Santos patches: send-dtmf.patch uploaded by Renato dos Santos (license #6267) Modified slightly for this commit for Asterisk 12. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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