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2008-10-31* Fixed timeout logic in the dialing API as setting timeoutsMark Michelson
had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-31Failover for func_odbc, allowing an INSERT query to be performed when the ↵Tilghman Lesher
UPDATE query initially affects 0 rows. (closes issue #13083) Reported by: Corydon76 Patches: 20081031__bug13083.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-30After seeing another problem in #asterisk stemming fromMark Michelson
the low default value of featuredigittimeout, I decided it was high time to change it. I have changed the default to 2000 ms based on a suggestion from Leif Madsen. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-30Pay attention to the searchcontexts entry in voicemail.conf (related to AST-125)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-23Thanks russellb for reminding an old man....Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-22Added debugging CLI functionsTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-18 Give app_authenticate the ability to select a prompt other than the default. BJ Weschke
(closes issue #13734) reported and patched by: jvandal git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-18 The QueueEntry event now has the uniqueid of the channel included.BJ Weschke
(closes issue #13731) reported and patched by: caio1982 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17Break up skinny.conf into seperate sections forMichiel van Baak
devices and lines. (closes issue #13412) Reported by: wedhorn Patches: config-restruct-v4.diff uploaded by wedhorn (license 30) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17Add an IAXregistry manager command. See doc/manager_1_1.txtMark Michelson
for more details of this command. (closes issue #13326) Reported by: ib2 Patches: bug13326_trunk_20080822.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-16support relative paths in musiconhold.conf, which makes moh work by default ↵Kevin P. Fleming
when Asterisk was configured using --prefix and 'make samples' is run git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14When specifying an invalid timeout to Dial, take itMark Michelson
to mean that no timeout is desired. (closes issue #13625) Reported by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-10Add keyword "same", which allows you to create multiple steps in a dialplan,Tilghman Lesher
without needing to respecify an extension pattern multiple times. (closes issue #13632) Reported by: blitzrage Patches: 20081006__bug13632.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage, Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09Add support for subscribing to a voice mailbox on a remote SIP server and ↵Joshua Colp
making the new/old message count available to local devices. (issue #AST-77) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-07fix wording as pointed out by CorydonMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06This commit introduces a change to how the "joinempty"Mark Michelson
and "leavewhenempty" options are configured in queues.conf. Instead of using vague terms like "yes," "no," "loose," and "strict," we now accept a comma-separated list of values to determine when to consider a member available. Extended details can be found in the queues.conf.sample file. Note also that the above four referenced values are still accepted for backwards-compatibility, but are mapped internally to the new method of representing the option. AST-105 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-03document meetme schedule changes (related to issue #11040)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-03put a note in CHANGES about the cli_cleanup done during AstriDevConMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-02The 'P' command for ExternalIVR was also added in 1.6.0Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-02TCP support for ExternalIVR went in to 1.6.1, not 1.6.0Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-02Permit the syntax and synopsis fields to be set (for func_odbc).Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-01tabs to spacesRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-30Add support for call pickup on Snom phones. Asterisk now includes a magicRussell Bryant
call-id in the dialog-info event package used with extension state subscriptions on Snom phones. Then, when the phone sends an INVITE with Replaces for the special callid, Asterisk will perform a pickup on the extension that was subscribed to. The original code on this issue was submitted by xylome. However, contributions have been made by (at least) mgernoth and pkempgen. The final patch was written by seanbright, and includes the necessary logic to allow this work in a technology independent way. (closes issue #5014) Reported by: xylome Patches: issue5014-trunk.diff uploaded by seanbright (license 71) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-10Move last change to CHANGES up to the 1.6.2 sectionRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-09Disable autoprune by default.Philippe Sultan
(closes issue #13411) Reported by: caio1982 Patches: res_jabber_autoprune1.diff uploaded by caio1982 (license 22) Tested by: caio1982 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-05Add the CURLOPT dialplan function, which permits setting various options forTilghman Lesher
use with the CURL dialplan function. (closes issue #12920) Reported by: davevg Patches: 20080904__bug12920.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, davevg git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-03Added 'skinny show lines verbose'Michiel van Baak
This will print the subs and their status for every line (if any). wedhorn did most of the work with his patch which introduced 'skinny show debug' but a discussion on IRC stated that it should be added to 'skinny show lines' Input on the output format by Qwell on IRC. (closes issue #13344) Reported by: wedhorn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-29Added the option s to the Park application which will silence the ↵Jeff Peeler
announcement of the parking space number. Also, fixes the bug of just clearing the flags instead of actually parsing the arguments to Park. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-26(closes issue #13366)Steve Murphy
Reported by: erousseau This was a reasonable enhancement request, which was easy to implement. Since it's an enhancement, it could only be applied to trunk. Basically, for accounting where "initiated" seconds are billed for, if the microseconds field on the end time is greater than the microseconds field for the answer time, add one second to the billsec field. The implementation was requested by erousseau, and I've implemented it as requested. I've updated the CHANGES, the cdr.conf.sample, and the .h files accordingly, to accept and set a flag for the corresponding new option. cdr.c adds in the extra second based on the usec fields if the option is set. Tested, seems to be working fine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-14Prepare for adding 1.6.2 changesRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-05Add '+=' append operator to configuration files.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-03Merge in changes that allow Asterisk to be built against the HoardSean Bright
memory allocator. See doc/hoard.txt for more details. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-01Merge changes from team/bbryant/keyrotationRussell Bryant
This set of changes enhances IAX2 encryption support by adding key rotation to provide enhanced security. The key used for encryption is rotated right after the call gets set up, and then again every few minutes. This was discussed at the last AstriDevCon. For interoperability with older versions of Asterisk, there is an option that disables key rotation. (closes issue #13018) Reported by: bbryant Patches: 07072008__iax2_key_rotation.diff uploaded by bbryant (license 36) Tested by: russell, bbryant git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-30Document adaptive capabilitiesTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-30Move implementation of an attended-transfer-complete sound from one channelTilghman Lesher
driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-28This commit compensates for buggy poll(2)Mark Michelson
implementations. Asterisk has, for a long time, had its own implementation of poll(2) which just used the input arguments to call select(2). In 1.4, this internal implementation was used for Darwin systems. This was removed in Asterisk trunk at some point, but it seems as though this was not the right move to make. On Mac OS X, it appears as though the poll used to gather CLI input does not respond properly when connecting via a remote Asterisk console. Reverting to the use of Asterisk's poll fixed the issue. Also, there is now an option for the configure script, --enable-internal-poll, which will allow for anyone to use Asterisk's internal poll implementation in case they suspect that their system's poll implementation is buggy. closes issue #11928) Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded by putnopvut (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-28Change SendImage() to output a more consistent status variable.Tilghman Lesher
(closes issue #13134) Reported by: eliel Patches: app_image.c.patch uploaded by eliel (license 64) UPGRADE.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-17Change several 'core' commands to be 'dialplan' commands (with appropriateTilghman Lesher
deprecation, of course) (closes issue #13016) Reported by: caio1982 Patches: dialplan_globals6.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-15Additional option for videosupport (always) that disables the optimization toTilghman Lesher
fail to setup video RTP if the two endpoints will not support it. This assists with call files and certain transfers to ensure that if two video phones are ever connected, they will always share a video feed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11clean up a bunch more Zaptel-related referencesKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-03Added a new option, "timeoutpriority" to queues.conf. A detailedMark Michelson
explanation of the change may be found in configs/queues.conf.sample (closes issue #12690) Reported by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-02The ackcall and endcall options in agents.conf now have supplemental optionsMark Michelson
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable instead of being hardcoded to '#' and '*'. (AST-86) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-26Improve consistency between app_dial and app_queue with regardsMark Michelson
to how language is handled between two channels whose native language is different. Prior to this patch, app_dial would have the callee inherit the caller's language, and app_queue would not. After this patch, app_dial no longer has the language inheritance capability. This seems to make the most sense since it seems more natural for a person to hear files played back in his/her native language instead of the language of the person on the far end of the call. See the CHANGES file for hints on how to keep the previous behavior of app_dial if desired. (closes issue #12489) Reported by: bcnit git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-24Update CHANGES and UPGRADE.txt per kpfleming's mail to #asterisk-dev.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-19OopsTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-19Allow alternative extensions to be specified for a user.Tilghman Lesher
(closes issue #12830) Reported by: jcollie Patches: astertisk-trunk-121496-alternate-extensions.patch uploaded by jcollie (license 412) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-17Changes to list peers and users in alpha. order, as per a reasonable request ↵Steve Murphy
in 12494. Due to changes in trunk to use the astobj2 i/f in the sip channel driver, the order of the entries in the config file was lost, thus the output was in a random order, but no longer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12Merged revisions 122127 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12Merged revisions 122046 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines (closes issue #10668) Reported by: arkadia Tested by: murf, arkadia Options added to forkCDR() app and the CDR() func to remove some roadblocks for CDR applications. The "show application ForkCDR" output was upgraded to more fully explain the inner workings of forkCDR. The A option was added to forkCDR to force the CDR system to NOT change the disposition on the original CDR, after the fork. This involves ast_cdr_answer, _busy, _failed, and so on. The T option was added to forkCDR to force obedience of the cdr LOCKED flag in the ast_cdr_end, all the disposition changing funcs (ast_cdr_answer, etc), and in the ast_cdr_setvar func. The CHANGES file was updated to explain ALL the new options added to satisfy this bug report (and some requests made verbally and via email, irc, etc, over the past months/year) The 's' option was added to the CDR() func, to force it to skip LOCKED cdr's in the chain. Again, the new options should be totally transparent to existing apps! Current behavior of CDR, forkCDR, and the rest of the CDR system should not change one little bit. Until you add the new options, at least! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-10Merge another big set of changes from team/russell/eventsRussell Bryant
This commit merges in the rest of the code needed to support distributed device state. There are two main parts to this commit. Core changes: - The device state handling in the core has been updated to understand device state across a cluster of Asterisk servers. Every time the state of a device changes, it looks at all of the device states on each node, and determines the aggregate device state. That resulting device state is what is provided to modules in Asterisk that take actions based on the state of a device. New module, res_ais: - A module has been written to facilitate the communication of events between nodes in a cluster of Asterisk servers. This module uses the SAForum AIS (Service Availability Forum Application Interface Specification) CLM and EVT services (Cluster Management and Event) to handle this task. This module currently supports sharing Voicemail MWI (Message Waiting Indication) and device state events between servers. It has been tested with openais, though other implementations of the spec do exist. For more information on testing distributed device state, see the following doc: - doc/distributed_devstate.txt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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