1174 Commits (2fd695d23c4e9e53e9aa7cba8c42067fa1a992b2)

Author	SHA1	Message	Date
Jeff Peeler	1172c38647	Add service maintenance message support ... This is the companion commit to libpri r732. Service messages are now supported for switch types 4ess/5ess. A new option service_message_support has been added to chan_dahdi.conf and is noted in the sample config file. The service message support is turned off by default. The current implementation relies on AstDB to keep track of channel state, which allows the statuses to be preserved across Asterisk restarts. Below is a description of the storage format. The state and reason for the service state are in the form <state>:<reason>, where: <state> ::= { 'O' } // 'O' – Out Of Service <reason> ::= { '0' | '1' | '2' | '3' }, where: '0' – No reason (backwards compatibility) '1' – NEAR END '2' – FAR END '3' – both NEAR and FAR END The new CLI commands to handle channel service state are: pri service disable channel <chan> pri service enable channel <chan> Many people contributed to the development of this functionality. Because I entered at the very end I do not know the exact history. Special thanks to all who moved the bug forward one way or another: cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7, tilghman, lmadsen, and especially dhubbard (he answered lots of my questions and did a large portion of the work) (closes issue #3450) Reported by: cmaj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2f048030bd	revert addition of LOG_SECURITY log channel; after further discussion, a much better solution will be used ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	4d74179f20	Add a new option, mwi_from, to sip.conf. ... This allows for you to change the From header for outgoing MWI NOTIFY requests. Prior to this, the best you could do was to set a callerid in the general section of sip.conf. The problem was that this was used for all outbound requests, not just MWI NOTIFY requests. AST-201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	b5f8c632df	add a dedicated log channel for modules to be able report security-related events, so that they can be fed into external processes for analysis and possible mitigation efforts ... (inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into Asterisk. ... The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	06061491ba	Merged revisions 186415 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	dababe2148	Merged revisions 186174 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. ... This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	08971ce205	Merged revisions 186059 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	9fd753a30e	Merged revisions 185121 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	da2230adf0	SIP preferred codec only feature ... Added an option to respond to a SIP invite with only the single most preferred joint codec. This limits the options of what codecs the other side can use. (closes issue #12485) Reported by: bamby Review: http://reviewboard.digium.com/r/206/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3fd19b3ab6	Merged revisions 183913 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	77a6840fd3	Add MFC/R2 support for chan_dahdi. ... This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	f1ae8e9f3b	Provide correct hint to debug SIP trouble in the default config ... (closes issue #14646) Reported by: strk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e69803a2be	Merged revisions 180380 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	3a14487abf	Allow for "magic" pickups to work when we wish to ignore the context ... When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	8970f8caaa	Merged revisions 180006 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	d2c5b0f1de	Mark res_ais as experimental, as the binary event format is subject to change. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Steve Murphy	ec6101595e	Merged revisions 178956 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	63561aea00	Sound confirmation of call pickup success. ... (closes issue #13826) Reported by: azielke Patches: pickupsound2-trunk.patch uploaded by azielke (license 548) __20081124_bug_13826_updated.patch uploaded by lmadsen (license 10) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	775ffb66d0	Clarifications on the different models and reference to further docs. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	fb540166d8	Merged revisions 178445 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	345a6fd1cb	Permit emailsubject and emailbody to be set per mailbox. ... (closes issue #14372) Reported by: fhackenberger Patches: voicemail_individual_subject_and_body_1.6.1 uploaded by fhackenberger (license 592) with additional fixes by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a1f583177e	ODBC transaction support ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. ... This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	176f380105	Typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	35ac1d7e1c	Fixed iax2 key rotation backwards compatibility ... Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. Review: http://reviewboard.digium.com/r/159/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	d11e6f0591	Add dynamic fax buffer configuration option to chan_dahdi.conf ... When the 'faxdetect' configuration option is used, one may also want to use the 'faxbuffers' configuration option in chan_dahdi.conf. This option will dynamically use the configured 'faxbuffers' buffer policy on a channel for the life of the call following the detection of fax tones. The faxbuffers buffer policy will be reverted during call teardown. An example use of 'faxbuffers' is below. This example would switch to using 6 buffers with a full buffer policy. faxbuffers=>6,full git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	178e6f06df	Adds force encryption option to iax.conf ... This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well. (closes issue #13285) Reported by: sgofferj Tested by: russell Review: http://reviewboard.digium.com/r/150/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	c15b83e7e5	Adds immediate yes/no option to iax.conf ... This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well. (closes issue #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk uploaded by jcovert (license 551) iax.conf.sample.patch uploaded by jcovert (license 551) Tested by: jcovert, dvossel Review: http://reviewboard.digium.com/r/143/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	69dff2f5f8	Update extensions.conf.sample to be correct. ... In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	673d85387a	Merged revisions 173070 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	fdcc0a9a60	Update the res_ldap.conf file with a better working example. ... (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	a010aa5ade	Remove incorrect line from sample config ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	8d782f96b8	Merged revisions 172517 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	97b4e9cf2a	channels/chan_dahdi.c ... * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b3ab95317c	Better document mode=multirow, based upon a conversation with Jared. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	0685c4b281	Update documentation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	aca43d126a	Add some more notes about device matching. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	2c4f19eb2c	Merged revisions 171837 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	131751140d	Make the sample skinny.conf work ... (closes issue #14325) Reported by: DEA Patches: skinny.conf.sample-trunk.txt uploaded by DEA (license 3) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	86f8225dfe	Merged revisions 170836 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines Remove superfluous implementation note (closes issue #14319) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	31f027a8c2	Merged revisions 170719 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan 2009) | 8 lines Add notes to the idlecheck explanation in res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000 Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	82d76adeb8	Add enhanced MWI generation to take advantage of new dahdi line reversal MWI ability. ... (closes issue #14104) Reported by: alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dbailey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	65120a3b33	Add discriminator for when ring pulse alert signal is used to preface MWI spills ... This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	5375047548	Merged revisions 168721 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 lines Meetme actually has realtime but wasn't documented ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	d4736e9897	Clarify some misunderstandings and make it even more clear that you can refer to a peer ... in the register= line. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	453b4cb8fb	Allow specifying a port number in the user portion of a register => line in sip.conf ... With this commit, a register => line in sip.conf may contain a port number in the "user" section of the line. Please see CHANGES and sip.conf.sample for more details regarding this. (closes issue #14198) Reported by: Nick_Lewis Patches: chan_sip.c-domainport2.patch uploaded by Nick (license 657) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f166220973	Merged revisions 168480 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines s/ringdance/ringcadence/ for Bulgaria ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	8969b03042	Update queues.conf.sample documentation. ... Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so. (closes issue #14179) Reported by: CrashHD Tested by: CrashHD git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago