asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Nicholson	49695230d3	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/branches/1.4@257070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	1d9be78f12	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/branches/1.4@255503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	5be8abf3d5	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/branches/1.4@253018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	8e30b3eafc	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/branches/1.4@252761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	c64fd68fd5	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/branches/1.4@252533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	d03a21d5f8	Revert last commit that had bad changed to configure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@252532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	0434ec7ad6	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/branches/1.4@252531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	529e8af144	Merged revisions 252089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 \| twilson \| 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) \| 20 lines 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/branches/1.4@252175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	19c43ed644	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/branches/1.4@250043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	793e58a924	Include examples of FILTER usage in extension patterns where a "." may be a risk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@245944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jared Smith	2594e9891a	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/branches/1.4@235181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	adc47c1cce	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/branches/1.4@233279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	2ef94c5196	Document a limitation in the AVAILSTATUS variable from ChanIsAvail and provide a workaround for it that does not change existing behavior. (closes issue #14426) Reported by: macli git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@229965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	ff7b512bcc	Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@226382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bedd6eb8a4	IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	050d830ec2	Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthias Nick	ebba623307	added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	96564de25e	Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d28e69ad5d	Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	05899c19a1	Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links before any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ed1951d895	Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	3300ac5b40	Clarify queues.conf comments to specify that variables should be set in the dialplan. (closes issue #15755) Reported by: trendboy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	361c9a99e1	Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	1e30dcf61c	Enhance configuration option for overlapdial allowing direction choice Previously overlap dialing could only be turned on or off for both incoming and outgoing calls. New parameters incoming, outgoing, and both have been added to allow further control. There is no change in default behavior with these new options and allows in band DTMF to be accepted in one direction if required. (closes issue #14471) Reported by: eboscani git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	5510a1c74e	error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b9004d067	Fix some spelling fail. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	58b5a85e80	Make absolute paths for logger channels work properly (Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6d08bad538	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/branches/1.4@186415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	09490bb688	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/branches/1.4@186174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	24fa699663	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/branches/1.4@186059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	63ca43071e	Update the channel allocation method documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	cf7131dd6a	Backport state interface changes to app_queue from trunk. After several issues raised on the Asterisk bugtracker against the 1.4 branch were determined to be fixable with the state interface change available in the 1.6.X series, it finally came time to just suck it up and backport the change. For a detailed explanation of what this change entails, the original trunk commit for this feature may be found here: http://svn.digium.com/view/asterisk?view=revision&revision=97203 In addition, the details for the use of this change to fix the problems stated in issue #12970 may be found in the review request I made for this change. It is linked below. (closes issue #12970) Reported by: edugs15 Review: http://reviewboard.digium.com/r/116 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	38934ec0d0	Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7e44582f57	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/branches/1.4@180380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ab5b88843c	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/branches/1.4@180006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	5996b58192	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/branches/1.4@178956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	48933e0c48	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/branches/1.4@178445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	13138151e1	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/branches/1.4@173070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	cefe4f025d	channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. These changes are already in trunk -r172400 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	2b340eab54	Rename new parkedcallparking option to parkedcallreparking Since this option actually already existed in 1.6.0+, use the same name so as not to confuse people when they upgrade git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	4e069885ce	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/branches/1.4@172517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	566429c300	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/branches/1.4@171837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3c19ad100a	Remove superfluous implementation note (closes issue #14319 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8fe9b2a600	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/branches/1.4@170719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	464b4a8a84	Meetme actually has realtime but wasn't documented git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5dca47a8bf	s/ringdance/ringcadence/ for Bulgaria git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a5ebe35d26	The documentation listed the ability to set 'maxmsg' per context. The truth is that you can only set this in the general section or per mailbox. Thus I am updating the sample config file to be more accurate. Thanks to sasargen on IRC for bringing up this issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@155011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	fffb7722be	A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	3f6d4154b8	Backport fix from 1.6.0 that allows you to set parkedcalltransfers=no\|caller\|callee\|both, but default to both which would be the equivalent of the existing behaviour. The problem was that if someone parked a call, the callee and caller would both get assigned the builtin transfer feature, which would not only be potentially giving someone the ability to transfer themselves when they shouldn't have it, but would also dissallow reinviting the media off of the call. (closes issue #12854) Reported by: davidw Patches: parkingfix4.diff.txt uploaded by otherwiseguy Tested by: davidw, otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@151763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
BJ Weschke	1d21453b49	An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c (closes issue #13709) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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896 Commits (91a8a41d8a99d69d6ab09ce55a7ba3983e23ad1b)