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2009-09-07Add docsoej1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@216655 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-27Merged revisions 203960 via svnmerge from russell1-8/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 | russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@203961 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16Merged revisions 200726 via svnmerge from kpfleming1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines Document the new automatic 'ignoresdpversion' behavior. Asterisk will now automatically ignore incorrect incoming SDP version numbers when necessary to complete a T.38 re-INVITE operation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@200727 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-16Add MFC/R2 support for chan_dahdi.russell1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10list the move of the astvarrundir from /var/run to /var/run/asteriskmvanbaak1-0/+4
(actually its $(localstatedir)/run/asterisk Makes setups with asterisk as non-root easier to manage because you can setup permissions on this dir instead of touching a file and setting permissions on that. Files that come to mind are asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180898 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merge phase 1 support for the new bridging architecture.file1-0/+3
This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180369 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Sound confirmation of call pickup success.tilghman1-0/+2
(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: http://svn.digium.com/svn/asterisk/trunk@178919 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Allows manager command to see if IAX link is trunked and encrypted. Displays ↵dvossel1-2/+6
what kind of encryption is enabled as well. Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not. (closes issue #14427) Reported by: snuffy Patches: iax_show_trunks.diff uploaded by snuffy (license 35) 2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review: http://reviewboard.digium.com/r/173/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178300 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-23Permit emailsubject and emailbody to be set per mailbox.tilghman1-0/+2
(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: http://svn.digium.com/svn/asterisk/trunk@178107 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-23list the addition of the SKINNY manager actions in the CHANGES file.mvanbaak1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178027 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-19ODBC transaction supporttilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177320 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-19Update CHANGES file to include MWI subscription support that was added some ↵file1-0/+2
time ago. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177291 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Merge queue-reset branch to Asteriskmmichelson1-0/+3
From a user point-of-view, this adds new CLI commands and Manager Actions to better facilitate the reloading of queues and the resetting of their statistics. The new CLI commands are the "queue reload" and "queue reset stats" commands. The new manager actions are the QueueReload and QueueReset commands. Review: http://reviewboard.digium.com/r/115 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175663 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13document G.722.1/.1C supportkpfleming1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175512 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13add 'faxbuffers' configuration option information to CHANGESdhubbard1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175475 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Adds force encryption option to iax.confdvossel1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@175344 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Adds immediate yes/no option to iax.confdvossel1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@174046 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Reverting commit number 173028 as there are somemmichelson1-2/+0
potential issues git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173047 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Add a CLI command to log out a manager usermmichelson1-0/+2
(closes issue #13877) Reported by: eliel Patches: cli_manager_logout.patch.txt uploaded by eliel (license 64) Tested by: eliel, putnopvut git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173028 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02This reverts the changes I made for 11583; willmurf1-2/+0
reviewboard this before committing again... reopened 11583 until all Russell's issues are resolved. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172929 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02This change allows the disconnect feature (as in "one-touch" in features.c)murf1-0/+2
to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172890 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-30Merged revisions 172517 via svnmerge from twilson1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@172580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Update documentationoej1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172270 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Yep. Documentation is important.oej1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@171925 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Adding AES_ENCRYPT and AES_DECRYPT dialplan functions. dvossel1-0/+4
(closes issue #14301) Reported by: amorsen review: http://reviewboard.digium.com/r/128/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@171757 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-16Fix a spelling mistake.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168760 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Related to issue #14246oej1-0/+2
Update changes for SIPRemoveHeader() git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168639 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-13Allow specifying a port number in the user portion of a register => line in ↵mmichelson1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@168575 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-09Add a script to find out the correct settings for Asterisk behind NATmvanbaak1-0/+3
(closes issue #13065) Reported by: tzafrir Patches: sip_nat_settings uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and moi git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168265 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-08Add the average talk time for a queuemmichelson1-2/+6
This patch adds the functionality to app_queue of calculating the average amount of time that channels are bridged for a queue. The algorithm used to calculate the average is the same exponential average currently used to calculate the average holdtime. See the CHANGES file to see the methods you may use to view this information. (closes issue #13960) Reported by: coolmig Patches: app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167792 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-08Convert dialplan application DAHDISendCallreroutingFacility to use commas.tilghman1-0/+2
(closes issue #13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167791 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-23Fix spelling error.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166625 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19Adding a new dialplan function AUDIOHOOK_INHERITmmichelson1-0/+2
This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166092 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Add a new application, Originate.russell1-0/+2
(closes issue #14075) Reported by: rcasas Patches: app_originate.c uploaded by rcasas (license 641), heavily modified by me Tested by: russell Review: http://reviewboard.digium.com/r/95/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165433 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17This patch adds a new 'ignoresdpversion' option to sip.conf. When this ismnicholson1-0/+8
enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165180 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Add timezone to the possible fields in a timespec.tilghman1-0/+8
(closes issue #14028) Reported by: mostyn Patches: timezone-v2.patch uploaded by mostyn (license 398) (with additional code guideline fixes and a memory leak fix by me - license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164976 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Qualify trumps poke per lmadsen.file1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164814 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Add configuration options for finer control over how Asterisk handles having ↵file1-0/+3
to poke all peers at seemingly the same time. (closes issue #13217) Reported by: cervajs git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164809 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Allow disabling pattern match searches within the Realtime dialplan switch.tilghman1-0/+2
(closes issue #13698) Reported by: fhackenberger Patches: 20081211__bug13698.diff.txt uploaded by Corydon76 (license 14) Tested by: fhackenberger git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164485 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-12Add a new CLI command, "channel redirect", which is similar in operationrussell1-0/+2
to AMI Redirect. Review: http://reviewboard.digium.com/r/89/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163716 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-08Add the ability to play a courtesy tone to the transfer target in a native ↵twilson1-0/+2
SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161679 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-04If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) ↵dhubbard1-0/+3
after T38 is negotiated. Terry Wilson created the original patch for this functionality, which I slightly modified and added the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not do anything for G711 over SIP fax detection. By default, this option is disabled. Reviewboard: http://reviewboard.digium.com/r/69/ This functionality is for issue AST-140. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161115 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-02Info on LOCAL_PEEK function.tilghman1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@160346 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-01Introduce CLI permissions.eliel1-0/+6
Based on cli_permissions.conf configuration file, we are able to permit or deny cli commands based on some patterns and the local user and group running rasterisk. (Sorry if I missed some of the testers). Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@160062 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-26add support for event suppression for AMI-over-HTTPkpfleming1-0/+15
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159629 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-25Add an option, waitfordialtone, for UK analog lines which do not end a calltilghman1-0/+5
until the originating line hangs up. (closes issue #12382) Reported by: one47 Patches: zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463) Tested by: fleed git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159317 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-21as suggested by jtodd, document the purposes of the CHANGES and UPGRADE fileskpfleming1-0/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@158449 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-19Commit CHANGES change I promised when submittingmmichelson1-0/+8
res_timing_timerfd git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157906 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-19Add info about REALTIME_FIELD and REALTIME_HASHtilghman1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157893 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-12This commit does two things:mvanbaak1-0/+2
- Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156120 f38db490-d61c-443f-a65b-d21fe96a405b