asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	15fd1e375c	Return error when no rows are deleted for AMI DBDelTree (closes issue AST-654) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6708ee76a0	Merged revisions 340219-340220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340219 \| twilson \| 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) \| 8 lines Add astdb conversion utility for Berkeley to SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10 they can use the astdb2bdb utility to convert the database back to the Berkeley format that Asterisk 1.8 uses. Review: https://reviewboard.asterisk.org/r/1502/ ........ r340220 \| twilson \| 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) \| 2 lines Add a missing file for the astdb2bdb conversion utility ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	2e2381341e	Clean up dsp.conf parsing Review: https://reviewboard.asterisk.org/r/1434/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	2d18de5f8f	Clean up cdr.conf parsing for [csv] section Review: https://reviewboard.asterisk.org/r/1427/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	61b369ac76	Clean up dnsmgr.conf parsing Review: https://reviewboard.asterisk.org/r/1432/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	404151ad65	New sip.conf option for setting default tonezone for channel or individual devices Review: https://reviewboard.asterisk.org/r/1429/ (closes issue ASTERISK-18497) Thanks to russellb for peer review. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	749ef800aa	Be more specific on which section has changed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	b52b026a35	Iterate though cdr.conf setting Review: https://reviewboard.asterisk.org/r/1426/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	052ece39ee	Merged revisions 332029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332029 \| mnicholson \| 2011-08-16 10:17:16 -0500 (Tue, 16 Aug 2011) \| 2 lines Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c3bd5892a6	Allow ENUM query functions to report lookup errors The ENUM dialplan functions do not report DNS query errors properly. It is useful to differentiate between failed query (e.g. non-existent domain) vs. no data records of the appropriate type. This is required to make overlapped dialing work. (closes issue ASTERISK-13769) Review: https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	16acfefa74	Merged revisions 331097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331097 \| twilson \| 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) \| 5 lines Bump the AMI protocol version to 1.2 As a result of converting Unlink events that were missed in the AMI 1.1 update to Bridge events, the AMI protocol version is being incremented. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	2c198555fd	Fix UPGRADE.txt files for Asterisk 10. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	1f65d55fb0	Merged revisions 328448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328448 \| lmadsen \| 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) \| 2 lines Update UPGRADE.txt and CHANGES files. Update documentation files stating that deprecated modules are no longer built by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	c98447f82c	Add UPGRADE-1.10.txt file from UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	13f92d2b82	Adds entry in UPDATES.txt for removal of formats/format_sln16.c. Fixes typo in CHANGES as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	efd040cd11	Replace Berkeley DB with SQLite 3 There were some bugs in the very ancient version of Berkeley DB that Asterisk used. Instead of spending the time tracking down the bugs in the Berkeley code we move to the much better documented SQLite 3. Conversion of the old astdb happens at runtime by running the included astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave identically to the old Berkeley backend, but in the future we could offer a much more robust interface. We do not include the SQLite 3 library in the source tree, but instead rely upon the distribution-provided libraries. SQLite is so ubiquitous that this should not place undue burden on administrators. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	f99a06d030	Commit "distrotech" app_queue changes to Trunk	15 years ago
Richard Mudgett	cdee44e992	Merged revisions 321337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert -r321331 and -r321332. ........ r321337 \| rmudgett \| 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) \| 7 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d211be98ed	Add note about PrivacyManager to UPGRADE.txt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	7a1204d129	Default to starting an autoservice in pbx_lua. The autoservice is automatically stopped when applications are executed, so this shouldn't cause any problems. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	d5e9ce9ab1	Make pbx_lua handle managing the autoservice better. Make autoservice_start() and autoservice_stop() return nothing. Also check if the autoservice flag is set before starting or stopping the autoservice and stop and start the autoservice when returning control to and getting control from the pbx engine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	6c38322870	Added note about changes in pbx_lua's behavior when applications do dialplan jumps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	2dfb427540	Add CEL extra field to cel_pgsql. (closes issue #18462) Reported by: joscas Patches: bug_18462.diff uploaded by snuffy (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	7f23115ad2	New HD ConfBridge conferencing application. Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	90177fe708	Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off hold. Added the moh_signaling option to specify what to do when the channel's bridged peer puts the ISDN channel on and off of hold. Implemented as a FSM to control libpri ISDN signaling when the bridged peer places the channel on and off of hold with the AST_CONTROL_HOLD and AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 Review: https://reviewboard.asterisk.org/r/1063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	5a28a27b0b	New CLI command 'gtalk show settings'. Review: https://reviewboard.asterisk.org/r/984/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	3162a8e558	Enable IPv6 for the built-in HTTP server. Review: https://reviewboard.asterisk.org/r/986 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c61b87c5f6	Shuffle UPGRADE.txt files for 1.10. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	a9e49f4e45	Update documentation for 'comebacktoorigin' in featuers.conf. The documentation for this option did not match the code. Fix that along with some minor cleanups to the code along the way. Document a slight change in behavior (to something that was previously undocumented) in UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char tag; }; struct ast_party_dialed { struct { char str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	745f4edbd5	Merged revisions 274280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274280 \| twilson \| 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) \| 9 lines Add option to not do a call forward on 482 Loop Detected Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ ........ (no option for trunk, just changing the behavior) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Bradley Latus	4405813297	Add High Resolution Times to CDRs for Asterisk People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change. Patch by snuffy. (closes issue #16559) Reported by: cianmaher Tested by: cianmaher, snuffy Review: https://reviewboard.asterisk.org/r/461/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	dfa82e0852	Update UPGRADE.txt and CHANGE for CDR functionality changes. Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity written unless cdr.conf exists and is configured. (closes issue #17373) Reported by: wdoekes Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	ffbb85bb4d	Set app and appdata fields when a Dial is redirected (closes issue #17204) Reported by: one47 Tested by: twilson, one47 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2da88f1977	Setup environment variables for the benefit of child processes and disallow changing them. (closes issue #14899) Reported by: jmls Patches: 20090916__issue14899.diff.txt uploaded by tilghman (license 14) Tested by: jmls git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	dd3343c33d	VoicemailMain and VMauthenticate, allow escape to the 'a' extension when a single '*' is entered Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber. This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape. If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour. Reported by: alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt uploaded by alecdavis (license 585) Review: https://reviewboard.asterisk.org/r/489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	bb2fa21ac1	IAXpeers output now matches SIPpeers format for manager (AMI). (closes issue #17100) Reported by: secesh Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/594/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	034a79c303	Updated doc for OSP lookup application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Ruggles	1649ae071c	ExternalIVR information for UPGRADE.txt added a paragraph about the fixes and changes to the ExternalIVR application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c207825dc7	Move an entry from CHANGES to UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2a1dce85b4	Move an entry from CHANGES that should be in UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2081809b07	update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by: miki git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	f62d03a8f3	Store the cause code that is returned when trying to create a channel in ChanIsAvail in the AVAILCAUSECODE dialplan variable instead of overwriting the device state in AVAILSTATUS. (closes issue #14426) Reported by: macli git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	6406f39594	DAHDI ISDN channel names will not allow device state to work. (Interim solution.) Since ISDN works like SIP and not analog ports in regard to devices, the device state based on the ISDN channel number could not work. This has not been an issue until the advent of PTMP NT mode. Previously, ISDN lines were used as trunks and did not have to keep track of specific devices. As an interim solution until device states are properly implemented, the channel name is being changed to the following format to use the generic device state support: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will work with the following restrictions: * The number of devices/phones cannot exceed the number of B channels. (i.e., BRI has 2) * Each device/phone can only have one number. No shared MSN's. * The phones/devices probably should not use subaddressing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	20743ec07d	Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	dd0c76a9d3	Move DAHDI/ISDN channel naming note from CHANGES to UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ff555b5e12	Sync up UPGRADE.txt with the 1.6.2 version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	17180120bf	Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	6f8e099b34	Merged revisions 218798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218798 \| russell \| 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) \| 9 lines Remove the IAXy firmware from Asterisk. The firmware can now be found on downloads.digium.com, where the rest of our binary downloads live. This was the last part of our Asterisk tarballs that was considered non-free by Debian. :-) (closes issue #15838) Reported by: paravoid ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	148552de24	Merged revisions 216085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216085 \| russell \| 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) \| 9 lines Merged revisions 216080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216080 \| russell \| 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) \| 2 lines Add a note about IAX2 to UPGRADE.txt. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

1 2 3 4 5

216 Commits (e2e6e511afa734e1e5d66b1084cfb266a31871e4)