asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	7c25fc6012	Prepare for adding 1.6.2 changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ff101d0b07	Add '+=' append operator to configuration files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6cf6d9eca5	Merge in changes that allow Asterisk to be built against the Hoard 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	17 years ago
Russell Bryant	58291bcec9	Merge changes from team/bbryant/keyrotation 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	17 years ago
Tilghman Lesher	01e189f6d8	Document adaptive capabilities git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	853f6a8b3e	Move implementation of an attended-transfer-complete sound from one channel 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	17 years ago
Mark Michelson	99db9f65b5	This commit compensates for buggy poll(2) 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	18 years ago
Tilghman Lesher	75d38f6024	Change SendImage() to output a more consistent status variable. (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	18 years ago
Tilghman Lesher	1517710d7e	Change several 'core' commands to be 'dialplan' commands (with appropriate 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	18 years ago
Tilghman Lesher	5a1d90e1fb	Additional option for videosupport (always) that disables the optimization to 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	18 years ago
Kevin P. Fleming	dd7630222c	clean up a bunch more Zaptel-related references git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	e4c93fc8c3	Added a new option, "timeoutpriority" to queues.conf. A detailed 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	18 years ago
Mark Michelson	953947b70b	The ackcall and endcall options in agents.conf now have supplemental options 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	18 years ago
Mark Michelson	0178d0ccd6	Improve consistency between app_dial and app_queue with regards 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	18 years ago
Sean Bright	00f74ac24c	Update CHANGES and UPGRADE.txt per kpfleming's mail to #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	2e0afd805b	Oops git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	122486b263	Allow alternative extensions to be specified for a user. (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	18 years ago
Steve Murphy	bb20ef7017	Changes to list peers and users in alpha. order, as per a reasonable request 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	18 years ago
Steve Murphy	86aaed2cc5	Merged revisions 122127 via svnmerge from 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	18 years ago
Steve Murphy	1cebe01dac	Merged revisions 122046 via svnmerge from 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	18 years ago
Russell Bryant	e9d72e0cb2	Merge another big set of changes from team/russell/events 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	18 years ago
Michiel van Baak	c5ea45af11	add a new argument to PrivacyManager to specify a context 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: https://origsvn.digium.com/svn/asterisk/trunk@121197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	07265a5033	Added a facility for sending arbitrary SIP notify commands from AMI. (closes issue #12562) Reported by: michael-fig Patches: 20080515__bug12562.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	1cebbfe268	Update CHANGES file for the things done in revision 120635. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	d81d206148	Adding two new queue log events. The ADDMEMBER event is logged when 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: https://origsvn.digium.com/svn/asterisk/trunk@120166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	c7191467d2	Add native AGI command GOSUB, as invoking Gosub with EXEC does not work 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: https://origsvn.digium.com/svn/asterisk/trunk@119296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e4d1b39bd8	Merged revisions 118646 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@118647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	975a848b67	A new feature thanks to the fine folks at Switchvox! 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: https://origsvn.digium.com/svn/asterisk/trunk@118173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	8f45823dda	add option 'a' to chanisavail. 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: https://origsvn.digium.com/svn/asterisk/trunk@118101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	ce8453f57c	Enhance ExternalIVR with new options and commands. (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: https://origsvn.digium.com/svn/asterisk/trunk@117725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	6353bddc57	Increase limit of unshared connections from 1023 to 4.2 billion. (Related to issue #12677) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	fced823c08	Change the default for the pridialplan parameter to the far more common case of '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: https://origsvn.digium.com/svn/asterisk/trunk@117182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	193d16cbde	Adding a new option to Chanspy(). The 'd' option allows for the spy to 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: https://origsvn.digium.com/svn/asterisk/trunk@116522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	bb386c84e7	Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss in text stream Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	29b1d73567	Add support for codec settings in originate via call file and manager. This is to enable video and text in originated calls. Development sponsored by Omnitor AB, Sweden. (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7daebcd610	Adding support for "urgent" voicemail messages. Messages which are 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: https://origsvn.digium.com/svn/asterisk/trunk@115588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	59817ce0d8	Update CHANGES file for previous commit of ENUM and TXCIDNAME changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	8b1d52c9a5	Allow a password change to be validated by an external script. (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: https://origsvn.digium.com/svn/asterisk/trunk@115582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	73581f3905	Optionally display the value of several variables within the Status command. (Closes issue AST-34) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	4f3e4e22ef	Add two new console commands "pri show version" and "ss7 show version" that will show the version of each library respectively. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	b5a127daac	Modify TIMEOUT() to be accurate down to the millisecond. (closes issue #10540) Reported by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	44af1e23d0	Merge changes from team/russell/smdi-msg-searching 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: https://origsvn.digium.com/svn/asterisk/trunk@115021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Brett Bryant	5634048c98	Add two new dialplan functions from libspeex for applying audio gain control 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: https://origsvn.digium.com/svn/asterisk/trunk@114926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	f4237076bf	Add support for specifying the registration expiry on a per registration basis in the register line. This comes from a Switchvox patch. (issue AST-24) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	e37dafdd3a	Adding new configuration options to app_queue. This adds two new values 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: https://origsvn.digium.com/svn/asterisk/trunk@114906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	fe2d50a4c9	Document the Incomplete application addition. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	3aad03e5f0	Adding a new option 'n' to app_chanspy. This option allows for the name of 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: https://origsvn.digium.com/svn/asterisk/trunk@114813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	d0f35e6355	Adding a new option, 'B' to app_chanspy. This option allows the spy to 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: https://origsvn.digium.com/svn/asterisk/trunk@114678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	01f3a08f8a	Add a c() option for the Jack() application and JACK_HOOK() funciton for supplying a custom client name. Using the channel name is still the default. This was done at the request of Jared Smith. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	c0b8f57b9d	(closes issue #12467 ) Reported by: atis Tested by: murf This upgrade adds the ~~ (concatenation) string operator to expr2. While not needed in normal runtime pbx operation, it is needed when raw exprs are being syntax checked. This plays into future syntax- unification plans. By permission of atis, this addition in trunk and the reason of why things are as they are will suffice to close this bug. I also added a short note about the previous addition of "sip show sched" to the CLI in CHANGES, which I discovered I forgot in a previous commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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337 Commits (33252a15471494ef7160db49fcc85097649c6228)