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2009-02-24Add section about the #exec command in configuration files.tilghman1-0/+8
(closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@178445 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-03Add warning to standard config, that globals may be overridden by othertilghman1-1/+7
dialplan configuration files. (closes issue #14388) Reported by: macli git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@173070 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02channels/chan_dahdi.crmudgett1-2/+4
* 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: http://svn.digium.com/svn/asterisk/branches/1.4@172962 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-31Rename new parkedcallparking option to parkedcallreparkingtwilson1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.4@172639 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-30Fix feature inheritance with builtin featurestwilson1-0/+6
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: http://svn.digium.com/svn/asterisk/branches/1.4@172517 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Add a better explanation of the difference between the device namespace and ↵oej1-0/+19
the dialplan for newbies. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@171837 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-24Remove superfluous implementation note (closes issue #14319)tilghman1-5/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@170836 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-23Add notes to the idlecheck explanation in res_odbc.conf.samplemmichelson1-0/+5
(closes issue #14319) Reported by: klaus3000 Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@170719 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Meetme actually has realtime but wasn't documentedoej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@168721 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-12s/ringdance/ringcadence/ for Bulgariarussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@168480 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-06The documentation listed the ability to set 'maxmsg' permmichelson1-2/+0
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: http://svn.digium.com/svn/asterisk/branches/1.4@155011 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-29A little documentation cross-ref between features andmurf1-5/+5
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: http://svn.digium.com/svn/asterisk/branches/1.4@152538 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-23Backport fix from 1.6.0 that allows you to set ↵twilson1-0/+2
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: http://svn.digium.com/svn/asterisk/branches/1.4@151763 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-15 An update to the documentation/example of agents.conf.sample with the ↵bweschke1-1/+1
correct parameter for this feature as defined in chan_agent.c (closes issue #13709) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@149683 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-30channels/chan_misdn.crmudgett1-3/+3
channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@145293 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create rules for disallowing contacts at certain addresses, which maytilghman1-0/+14
improve the security of various installations. As this does not change any default behavior, it is not classified as a direct security fix for anything within Asterisk, but may help PBX admins better secure their SIP servers. (closes issue #11776) Reported by: ibc Patches: 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@142865 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-15More fixes for realtime peers.tilghman1-1/+3
(closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@138258 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-14Comments in this config file were aligned only if your tab size was set to 8.russell1-334/+334
So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@137731 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06* The allowed_bearers setting in misdn.conf misspelled onermudgett1-4/+13
of its options: digital_restricted. * Fixed some other spelling errors and typos. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@136241 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-04fix a config sample typorussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@135536 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-04Add a minor clarification to the documentation of mohinterpret and mohsuggestrussell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@135473 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-22Merged revisions 132711 via svnmerge from tilghman1-0/+10
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@132713 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-22use renamed libpri API call for controlling this feature (was improperly ↵kpfleming1-1/+1
named before) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@132641 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11new installations should be using DAHDI instead of Zaptel, so the sample ↵kpfleming5-19/+19
config file is now chan_dahdi.conf instead of zapata.conf also, convert remaining references to zapata.conf in various places git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@130042 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11add support for a configuration parameter for 'inband audio during RELEASE', ↵kpfleming1-0/+4
which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today (related to issue #13042) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@130039 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01Clear up documentation on "domain=" setting in sip.confoej1-4/+3
Reported by: davidw (closes issue #12413) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@126844 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-30Add note about other names for EuroISDNtilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@126674 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-26Document ackcall=always.tilghman1-0/+2
(closes issue #12852) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@125218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-19Correct description of notifyringing option.tilghman1-1/+2
(Closes issue #12890) Reported by gminet git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@123883 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Add an option to use the source IP address of RTP as the destination IP ↵file1-1/+6
address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@118646 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-27Add a note that pbx_config.so is needed for Local channels.tilghman1-1/+4
(Closes issue #12671) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@118358 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-14Document exitcontext in app_voicemail sample configqwell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@116409 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-25Reference documentation files that actually exist.tilghman4-5/+5
(closes issue #12516) Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded by linuxmaniac (license 472) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114649 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-09If the [csv] section does not exist in cdr.conf, then an unload/load sequencetilghman1-2/+2
is needed to correct the problem. Track whether the load succeeded with a variable, so we can fix this with a simple reload event, instead. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@113874 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-07Allow playback with noanswer (and add earlyrtp option).qwell1-0/+5
(closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@113118 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-26Merge changes from team/russell/smdi-1.4russell1-0/+32
This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@104119 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-11improve 2BCT documentation a bit (thanks Jared)kpfleming1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@103315 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-07document usage of 'transfer' configuration option for ISDN PRI switch-side ↵kpfleming1-2/+6
transfers git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@102807 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-06Clarify setting DYNAMIC_FEATURES so that it gets inherited by outbound channels.russell1-1/+4
(due to a discussion between me and a user via email) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@102651 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-30Change default config to use descending channel order of groups, rather than ↵qwell1-4/+4
ascending. Fixes a potential source of confusion in glare-type situations. Issue 11875, reported by JimVanM. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@101219 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-21Permit the user to specify number of seconds that a connection may remain idle,tilghman1-0/+5
which fixes a crash on reconnect with the MyODBC driver. (closes issue #11798) Reported by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@99341 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17doh! revert a revert of a revert (changed by mistake in 99010)pari1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@99014 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17missed that one while revertingpari1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@99010 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17Add a clarification about the immediate= option of zapata.confqwell1-0/+2
Issue 11784, patch by klaus3000. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@98991 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Remove other remnants of pbx_kdeconsolerussell1-3/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@97753 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07Merged revisions 96931 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@96932 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28it is impossible to set permissions for manager accounts created by ↵kpfleming1-0/+5
users.conf (reported internally, patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@90098 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Add a note to the sample voicemail config noting that when using IMAP storage,russell1-1/+2
only the first format specified will be attached to the message. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89634 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Clarify limitonpeers=yesoej1-0/+3
(closes issue #11304) Reported by: pj git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89624 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27closes issue #11379; OK, this is an attempt to make both sides happy. To the ↵murf1-0/+10
cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89622 f38db490-d61c-443f-a65b-d21fe96a405b
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