1322 Commits (7eac4cfb03feaede698b6326c64543a4ac087a06)

Author	SHA1	Message	Date
Tilghman Lesher	e940ef8c4c	Logic fixups for a sample FREENUM dialplan context. ... (closes issue #17263) Reported by: pprindeville Patches: freenum-dialplan.patch#3 uploaded by pprindeville (license 347) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4532cd2464	Pattern match fail. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	3e04d6fe8e	Merged revisions 259270 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	ea9186d4ea	Add 'soft hangup' alias per Steve Johnson on asterisk-users. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	a8aef91e9d	Add example dialplan for dialing ISN numbers (http://www.freenum.org). ... Minor tweaks and documentation added by me. (closes issue #17058) Reported by: pprindeville Patches: freenum.patch#5 uploaded by pprindeville (license 347) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3bb60ae5b7	Removing unused configuration parameters ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8b7a90a026	Yet another issue where the conversion of the application delimiter to comma caused an issue. ... Application arguments within the feature map could possibly contain a comma, which conflicts with the syntax of the features.conf configuration file. This patch allows the argument to be wrapped in parentheses or quoted, to allow the application arguments to be interpreted as a single configuration parameter. (closes issue #16646) Reported by: pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2724f89bba	Merged revisions 257070 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines Add an option to restore past broken behavor of the Events manager action Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested. (closes issue #17023) Reported by: nblasgen Review: https://reviewboard.asterisk.org/r/602/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b1abf9234f	Update sample dialstrings in sip.conf.sample file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. ... From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	ad3af59345	Removed documentation of the non existent 'both' option to 'faxdetect' in sip.conf ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	2de9cd0d38	Add documentation clarifying when 't' and 'T' can be used. ... (closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	8a3576d16c	Replace some documentation from 1.6.x back into trunk. ... This documentation associated wth tlsbindaddr is still useful so lets synchronize it between trunk and 1.6.x branches. (issue #17054) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	aae9d51510	Update confusing documentation for tlsbindaddr. ... Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	42577406fd	Improve handling of T.38 re-INVITEs that arrive before a T.38-capable ... application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	560d5c6099	Allow configuration of minsecs and nextaftercmd per mailbox. ... Previously only configurable globally. A unit test has also been written to provide protection against parse failures for supported mailbox options. (closes issue #16864) Reported by: kobaz Patches: voicemail2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/555/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	589c397065	Accomodate equal signs in DSNs and add documentation, based upon mmichelson's feedback. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	13397e9446	Merged revisions 253018 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) | 6 lines Add french snipset to say.conf. Add the french snipset to say.conf. (Closes issue #15799) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	26d2f39520	Merged revisions 252761 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines Additional extensions.ael global variable fixes. Fixing up a couple more overlapping global variable namespaces shared with extensions.conf.sample. Also noticed a few of the lines that were commented out didn't have the closing semi-colon so I added that as well. (issue #17035) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	2b9d9f8ec8	Merged revisions 252533 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines Update extensions.ael file to not overlap extensions.conf. Updated the extensions.ael file so the global variables don't overlap those that we have in extensions.conf (sample files). This way unexpected things won't happed hopefully if both pbx_ael and res_config are loaded. (closes issue #17035) Reported by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alexandr Anikin	fa9d6969d6	generate roundtrip delay requests and responses ... added response to roundtrip delay requests from opposite side added roundtrip delay request sending to opposite side after answer, added options for sending request (interval between request and count of unreplied requests before forced call hangup) (closes issue #16976) Reported by: vmikhelson Patches: rtdr-1.6.0-2.patch uploaded by may213 (license 454) Tested by: vmikhelson, may213 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	68d1ded8dd	Only change the RTP ssrc when we see that it has changed ... This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6bd57e0720	Add new config option to control AMI alarm event reporting in chan_dahdi. ... New config parameter "reportalarms" added in chan_dahdi.conf which supports the following possible values: "channels": report each channel alarms (current behavior, default for backward compatibility) "spans": report an "SpanAlarm" event when the span of any configured channel is alarmed "all": report channel and span alarms (aggregated behavior) "none": do not report any alarms (closes issue #16709) Reported by: nahuelgreco Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	49aaeb1df2	Merge missed files from res_fax/res_fax_spandsp merge. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	b838b15702	Merged revisions 250043 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines Update documentation to clarify purpose of unanswered option. (closes issue #16267) Reported by: elsto Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested by: davidw, elsto ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	862ebf4d00	fixes adaptive jitterbuffer configuration ... When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	57c819fd5e	addition of dynamic parkinglots feature ... This feature allows for parkinglots to be created dynamically within the dialplan. Thanks to all who were involved with getting this patch written and tested! (closes issue #15135) Reported by: IgorG Patches: features.dynamic_park.v3.diff uploaded by IgorG (license 20) 2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7) dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested by: eliel, IgorG, acunningham, mvanbaak, zktech Review: https://reviewboard.asterisk.org/r/352/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5b86e43b30	Merged revisions 245944 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines Include examples of FILTER usage in extension patterns where a "." may be a risk. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	38cb3e2ac9	Remove useless sip options related to hash table size. ... First off, these options weren't actually doing anything. By the time the options were parsed, the peer and dialog containers had already been allocated with their default values. Second, hash table size is something that doesn't really make sense to change in a config file. If a user is that interested in changing the hashtable size, he can modify the source itself. I have removed the parsing of the hash_peer, hash_user, and hash_dialog options. I have removed the hash_user_size variable altogether since it is not used at all. I also changed hash_peer_size and hash_dialog_size to be constant, and have changed the symbols to be in all caps as constants typically are. I have also removed the entire section in sip.conf.sample regarding configurable hashtable sizes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	1ef8082cd3	Clarify RTP NAT handling a bit. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	f44630cd01	Add more examples to extensions.conf showing how to use various ... functionality and provide commonly useful features. (closes issue #16090) Reported by: pprindeville Patches: extensions.conf-bugid16090.patch#3 uploaded by pprindeville (license 347) Tested by: tzafrir, pprindeville, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	c2f486e118	Note that direct T.38 is not supported. ... (closes issue #16411) Reported by: stanusr Patches: __20091210-sip.conf.sample-documentation.txt uploaded by lmadsen (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b4504ada8b	Permit dates and times to be stored in timezones other than the default (typically, UTC) ... (closes issue #16401) Reported by: lordmortis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	2aceadfa82	Document the usefulness of explicit udp:// in the register string ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	28d16a3cb1	1. Updated for OSP Toolkit 3.6.0. ... 2. Added service type ported number query. 3. Formated code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6b34563778	Add auth_policy option to jabber.conf for auto user registration. ... The option is global and currently the acceptable values as noted in the sample config are accept or deny. (closes issue #15228) Reported by: lp0 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jared Smith	fb931dac4f	Merged revisions 235181 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15 Dec 2009) | 4 lines Add a line showing that we can use CIDR notation. patch by jsmith, after discussion with jtodd ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	89a1af1d38	Allow greetings-only mailboxes for Voicemail. ... (closes issue #15132) Reported by: floletarmo Patches: voicemail_changes.patch uploaded by floletarmo (license 784) (with some additional changes by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	4fd45889df	Merged revisions 233279 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines clarify requirecalltoken option in iax.sample.conf (closes issue #16223) Reported by: bklang Patches: clarify-iax-requirecalltoken.patch uploaded by bklang (license 919) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e21deabf02	new queue option, penaltymemberslimit, disregards penalty on too few queue members when enabled ...	17 years ago
Tilghman Lesher	8599628e0b	Add pagerdateformat, to allow shorter dates for SMS messages. ... (closes issue #16263) Reported by: andrew Patches: pagerdate.patch uploaded by andrew (license 240) (with a slight modification by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e5aa8cad9b	Extend voicemail to allow IMAP folders to be specified per mailbox. ... Previously only possible per context, new option called imapfolder. (closes issue #14298) Reported by: jablko Patches: patch-200906202 uploaded by jablko (license 675) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	60e10aba46	Change fax detection in chan_sip so it behaves as one would expect. ... Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5e2aa190fe	Display a list of channel variables in each channel-oriented event. ... (Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	75c015bfff	Add the capability to require a module to be loaded, or else Asterisk exits. ... Review: https://reviewboard.asterisk.org/r/426/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	33fedbdb54	Update sample config for ALSA mute and noaudiocapture ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229754 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	e7c7dac8a9	Update sip.conf.sample. ... Just updating a spelling error and some capitalization in a documentation update that Olle added. May the Swenglish be with you. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	8e583db28f	Clarification ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	cca751350a	Clarify some security issues early in the sample configuration ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	aabff54c4b	Add the 'relative-periodic-announce' option to app_queue to allow for calculating the time of announcments from the end of the previous announcment rather than from the beginning. ... (closes issue #15260) Reported by: tonils git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago