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2008-08-29Added the option s to the Park application which will silence the ↵jpeeler1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@140491 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-26(closes issue #13366)murf1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@140057 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-14Prepare for adding 1.6.2 changesrussell1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137901 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05Add '+=' append operator to configuration files.tilghman1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135717 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-03Merge in changes that allow Asterisk to be built against the Hoardseanbright1-0/+2
memory allocator. See doc/hoard.txt for more details. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135405 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-01Merge changes from team/bbryant/keyrotationrussell1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@135158 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Document adaptive capabilitiestilghman1-0/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134443 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Move implementation of an attended-transfer-complete sound from one channeltilghman1-2/+7
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: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28This commit compensates for buggy poll(2)mmichelson1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@134125 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28Change SendImage() to output a more consistent status variable.tilghman1-0/+4
(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: http://svn.digium.com/svn/asterisk/trunk@134088 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-17Change several 'core' commands to be 'dialplan' commands (with appropriatetilghman1-0/+5
deprecation, of course) (closes issue #13016) Reported by: caio1982 Patches: dialplan_globals6.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131606 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-15Additional option for videosupport (always) that disables the optimization totilghman1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@130951 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11clean up a bunch more Zaptel-related referenceskpfleming1-9/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130044 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-03Added a new option, "timeoutpriority" to queues.conf. A detailedmmichelson1-0/+3
explanation of the change may be found in configs/queues.conf.sample (closes issue #12690) Reported by: atis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127720 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-02The ackcall and endcall options in agents.conf now have supplemental optionsmmichelson1-0/+7
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable instead of being hardcoded to '#' and '*'. (AST-86) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127558 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-26Improve consistency between app_dial and app_queue with regardsmmichelson1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@125647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-24Update CHANGES and UPGRADE.txt per kpfleming's mail to #asterisk-dev.seanbright1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124835 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-19Oopstilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124125 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-19Allow alternative extensions to be specified for a user.tilghman1-0/+2
(closes issue #12830) Reported by: jcollie Patches: astertisk-trunk-121496-alternate-extensions.patch uploaded by jcollie (license 412) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124049 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-17Changes to list peers and users in alpha. order, as per a reasonable request ↵murf1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@123448 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12Merged revisions 122127 via svnmerge from murf1-2/+1
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: http://svn.digium.com/svn/asterisk/trunk@122128 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12Merged revisions 122046 via svnmerge from murf1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@122091 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-10Merge another big set of changes from team/russell/eventsrussell1-20/+31
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: http://svn.digium.com/svn/asterisk/trunk@121559 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-08add a new argument to PrivacyManager to specify a contextmvanbaak1-0/+3
where the entered phone number is checked. You can now define a set of extensions/exten patterns that describe valid phone numbers. PrivacyManager will check that context for a match with the given phone number. This way you get better control. For example people blindly hitting 10 digits just to get past privacymanager Example line in extensions.conf: exten => incoming,n,PrivacyManager(3,10,,route-outgoing) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121197 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-06Added a facility for sending arbitrary SIP notify commands from AMI.tilghman1-0/+1
(closes issue #12562) Reported by: michael-fig Patches: 20080515__bug12562.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121042 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-05Update CHANGES file for the things done in revision 120635.bbryant1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120673 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-03Adding two new queue log events. The ADDMEMBER event is logged whenmmichelson1-0/+5
a dynamic realtime queue member is added to the queue, and the REMOVEMEMBER event is logged when a dynamic realtime member is removed. Since no calling channel is associated with these events the string "REALTIME" is placed where the channel's unique id is normally placed. (closes issue #12774) Reported by: atis Patches: queue_log_rt_members.patch uploaded by atis (license 242) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120166 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-30Add native AGI command GOSUB, as invoking Gosub with EXEC does not worktilghman1-0/+3
properly. (closes issue #12760) Reported by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119296 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-28Merged revisions 118646 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP 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/trunk@118647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-23A new feature thanks to the fine folks at Switchvox!mmichelson1-0/+6
If a deadlock is detected, then the typical lock information will be printed along with a backtrace of the stack for the offending threads. Use of this requires compiling with DETECT_DEADLOCKS and having glibc installed. Furthermore, issuing the "core show locks" CLI command will print the normal lock information as well as a backtraces for each lock. This requires that DEBUG_THREADS is enabled and that glibc is installed. All the backtrace features may be disabled by running the configure script with --without-execinfo as an argument git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118173 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-23add option 'a' to chanisavail.mvanbaak1-0/+2
If you give chanisavail a list of channels, it will only return the first available channel. When this option is set, it will return all the available channels from the given list. (closes issue #12248) Reported by: dagmoller Patches: app_chanisavail-snv.patch-v2.txt uploaded by dagmoller (license 436) - major changes by me because russellb pointed out some buffer overflows and codeguideline issues. Converted it all to the ast_str_* api Tested by: dagmoller, mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118101 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22Enhance ExternalIVR with new options and commands.tilghman1-0/+3
(closes issue #12705) Reported by: ctooley Patches: new_externalivr_argument_format-v2.diff uploaded by ctooley (license 136) new_externalivr_documentation.diff uploaded by ctooley (license 136) and a few additional fixes by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117725 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-20Increase limit of unshared connections from 1023 to 4.2 billion.tilghman1-10/+13
(Related to issue #12677) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117264 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-19Change the default for the pridialplan parameter to the far more common case oftilghman1-0/+1
'unknown', and better document the use of each parameter. (closes issue #12633) Reported by: tzafrir Patches: pridialplan_unknown_2.diff uploaded by tzafrir (license 46) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117182 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-14Adding a new option to Chanspy(). The 'd' option allows for the spy tommichelson1-0/+5
press DTMF digits to switch between spying modes. Pressing 4 activates spy mode, pressing 5 activates whisper mode, and pressing 6 activates barge mode. Use of this feature overrides the normal operation of DTMF numbers. This feature is courtesy of Switchvox. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116522 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-14Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss ↵oej1-0/+2
in text stream Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116237 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-14Add support for codec settings in originate via call file and manager.oej1-0/+1
This is to enable video and text in originated calls. Development sponsored by Omnitor AB, Sweden. (http://www.omnitor.se) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116229 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-09Adding support for "urgent" voicemail messages. Messages which aremmichelson1-0/+6
marked "urgent" are considered to be higher priority than other messages and so they will be played before any other messages in a user's mailbox. There are two ways to leave an urgent message. 1. send the 'U' option to VoiceMail(). 2. Set review=yes in voicemail.conf. This will give instructions for a caller to mark a message as urgent after the message has been recorded. I have tested that this works correctly with file and ODBC storage, and James Rothenberger (who wrote initial support for this feature) has tested its use with IMAP storage. (closes issue #11817) Reported by: jaroth Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent Tested by: putnopvut, jaroth git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115588 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-09Update CHANGES file for previous commit of ENUM and TXCIDNAME changes.bbryant1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115586 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-09Allow a password change to be validated by an external script.tilghman1-0/+5
(closes issue #12090) Reported by: jaroth Patches: vm-check-newpassword.diff.txt uploaded by mvanbaak (license 7) 20080509__bug12090.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115582 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-05Optionally display the value of several variables within the Status command.tilghman1-0/+5
(Closes issue AST-34) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115301 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Add two new console commands "pri show version" and "ss7 show version" that ↵bbryant1-0/+5
will show the version of each library respectively. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115078 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Modify TIMEOUT() to be accurate down to the millisecond.tilghman1-0/+4
(closes issue #10540) Reported by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115076 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Merge changes from team/russell/smdi-msg-searchingrussell1-0/+4
This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function. Previously, this function only allowed searching by the forwarding station. I have added some options to allow you to also search for messages in the queue by the message desk terminal ID, as well as the message desk number. This originally came up as a suggestion on the asterisk-dev mailing list. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115021 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Add two new dialplan functions from libspeex for applying audio gain control bbryant1-0/+3
and denoising to a channel, AGC() and DENOISE(). Also included, is a change to the audiohook API to add a new function (ast_audiohook_remove) that can remove an audiohook from a channel before it is detached. This code is based on a contribution from Switchvox. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114926 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-30Add support for specifying the registration expiry on a per registration ↵file1-0/+2
basis in the register line. This comes from a Switchvox patch. (issue AST-24) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114912 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-30Adding new configuration options to app_queue. This adds two new valuesmmichelson1-0/+6
to announce-position, "limit" and "more," as well as a new option, announce-position-limit. For more information on the use of these options, see CHANGES or configs/queues.conf.sample. (closes issue #10991) Reported by: slavon Patches: app_q.diff uploaded by slavon (license 288) Tested by: slavon, putnopvut git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114906 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-30Document the Incomplete application addition.tilghman1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114874 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-28Adding a new option 'n' to app_chanspy. This option allows for the name of ↵mmichelson1-0/+4
the spied-on party to be spoken instead of the channel name or number. This was accomplished by adding a new function pointer to point to a function in app_voicemail which retrieves the name file and plays it. This makes for an easy way that applications may play a user's name should it be necessary. app_directory, in particular, can be simplified greatly by this change. This change comes as a suggestion from Switchvox, which already has this feature. AST-23 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114813 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-25Adding a new option, 'B' to app_chanspy. This option allows the spy tommichelson1-0/+3
barge on the call. It is like the existing whisper option, except that it allows the spy to talk to both sides of the conversation on which he is spying. This feature has existed in Switchvox, and this merges the functionality into Asterisk. (AST-32) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114678 f38db490-d61c-443f-a65b-d21fe96a405b