- Jan 06, 2014
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Richard Mudgett authored
* The core external MWI resource provides for MWI message counts persistence using sorcery. With sorcery, the user is able to configure which sorcery wizzard backend to use if the default astdb is not desired. * The core external MWI resoruce provides some debugging CLI commands enabled by defining MWI_DEBUG_CLI. The debugging CLI commands are: "mwi delete all", "mwi delete like <regex>", "mwi delete mailbox <mailbox>", "mwi list all", "mwi list like <regex>", "mwi show mailbox <mailbox>", and "mwi update mailbox <mailbox> [<new> [<old>]]". (closes issue AFS-43) Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged revisions 404952 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 02, 2014
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Kevin Harwell authored
Added a new 'set_var' option for ast_sip_endpoint(s). For each variable specified that variable gets set upon creation of a pjsip channel involving the endpoint. (closes issue ASTERISK-22868) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3095/ ........ Merged revisions 404663 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 23, 2013
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Tzafrir Cohen authored
If ignore_failed_channels is set to "true" for a channel, the channel will continue to be configured even if configuring it has failed. This allows Asterisk to start before all the DAHDI initialization is done and thus not force the starting order dahdi -> asterisk. Review: https://reviewboard.asterisk.org/r/3063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 20, 2013
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Rusty Newton authored
Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity. Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf. I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options. (issue ASTERISK-23004) (closes issue ASTERISK-23004) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3086/ ........ Merged revisions 404405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 19, 2013
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Richard Mudgett authored
This change is in preparation for external MWI support. Removed code from the system for normal mailbox handling that appends @default to the mailbox identifier if it does not have a context. The only exception is the legacy hasvoicemail users.conf option. The legacy option will only work for app_voicemail mailboxes. The system cannot make any assumptions about the format of the mailbox identifer used by app_voicemail. chan_sip and chan_dahdi/sig_pri had the most changes because they both tried to interpret the mailbox identifier. chan_sip just stored and compared the two components. chan_dahdi actually used the box information. The ISDN MWI support configuration options had to be reworked because chan_dahdi was parsing the box@context format to get the box number. As a result the mwi_vm_boxes chan_dahdi.conf option was added and is documented in the chan_dahdi.conf.sample file. Review: https://reviewboard.asterisk.org/r/3072/ ........ Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 18, 2013
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Alexandr Anikin authored
Introduce new config option 'aniasdni'. If yes then asterisk set dialed number as own id back to the caller on incoming h.323 calls. Option can be set globally or per user section. (closes issue ASTERISK-22020) Reported by: Ross Beer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 17, 2013
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David M. Lee authored
........ Merged revisions 404006 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 16, 2013
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David M. Lee authored
This patch allows individual dialplan functions to be marked as 'dangerous', to inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. Also, the ABI was changed to something more reasonable, since Asterisk 12 does not yet have a public release. (closes issue ASTERISK-22905) Review: http://reviewboard.digium.internal/r/432/ ........ Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 11, 2013
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Kevin Harwell authored
In some cases messages need to be sent to a direct URI (sip:<ip address>). This patch adds in that support by using a default outbound endpoint. When sending messages, if no endpoint can be found then the default one is used. To facilitate this a new default_outbound_endpoint option was added to the globals section for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/ ........ Merged revisions 403680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 22, 2013
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Kevin Harwell authored
Renamed, where appropriate, the configuration options for chan/res_pjsip to use snake case (compound words separated by an underscore). For example, faxdetect will become fax_detect, recordofffeature will become record_off_feature, etc... Review: https://reviewboard.asterisk.org/r/3002/ ........ Merged revisions 403022 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 15, 2013
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Jonathan Rose authored
Review: https://reviewboard.asterisk.org/r/3008/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 08, 2013
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Jonathan Rose authored
Security Events will now be written to any listener of the new 'security' class Review: https://reviewboard.asterisk.org/r/2998/ ........ Merged revisions 402584 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 02, 2013
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Richard Mudgett authored
ConfBridge allows custom DTMF menus to be created in the confbridge.conf file by assigning a DTMF key sequence to a sequence of actions as follows: DTMF-sequence = action,action... Unfortunately, the normal config file processing code interprets an initial '#' character as starting a directive such as #include. * Add the ability to escape the first non-blank character in a config line so the '#' character can be used without triggering the directive processing code. (closes issue AFS-2) (closes issue ASTERISK-22478) Reported by: Nicolas Tanski Patches: jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified) Review: https://reviewboard.asterisk.org/r/2969/ ........ Merged revisions 402407 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402416 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 23, 2013
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Michael L. Young authored
When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses was removed by mistake. This patch restores this functionality and adds notes about IPv6 addresses in the sample config. (closes issue ASTERISK-22741) Reported by: Joshua Colp Tested by: Michael L. Young Patches: asterisk-22741-fix-binding-multiple-addr.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2945/ ........ Merged revisions 401488 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 14, 2013
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Richard Mudgett authored
* Add hwtxgain and hwrxgain config options to chan_dahdi.conf with documentation in chan_dahdi.conf.sample. (closes issue ASTERISK-22429) Reported by: Jaco Kroon Patches: jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch uploaded by rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 08, 2013
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Richard Mudgett authored
ConfBridge now has the ability to set the language of announcements to the conference. The language can be set on a bridge profile in confbridge.conf or by the dialplan function CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983) Reported by: Jonathan White Patches: M19983_rev2.diff (license #5138) patch uploaded by junky (modified) Tested by: rmudgett ........ Merged revisions 400741 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400742 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 04, 2013
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Jonathan Rose authored
Also updates sample configurations for sorcery and extconfig to demonstrate how to use databases created by that alembic script. (closes issue ASTERISK-22133) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2892/ ........ Merged revisions 400532 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Sep 30, 2013
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Jonathan Rose authored
This patch also adds documentation for parking from features.conf to res_parking.conf ........ Merged revisions 400205 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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David M. Lee authored
........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Kinsey Moore authored
This adds a global option in chan_sip to allow it to continue attempting registration if a 403 is received, clearing the cached nonce and treating it as a non-fatal response. Normally, this would cause registration attempts to that endpoint to stop. This also adds a similar per-outbound-registration option to chan_pjsip which allows the retry interval to be altered for 403 responses to REGISTER requests. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi ........ Merged revisions 400137 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400140 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400141 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Sep 28, 2013
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Matthew Jordan authored
We forgot to add documentation for res_pjsip_notify, which would prevent it from being loaded. Whoops. This patch also updates res_pjsip_notify to use pjsip_notify.conf, which now has its own sample file in the configs directory as well. Review: https://reviewboard.asterisk.org/r/2835/ ........ Merged revisions 400121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Sep 27, 2013
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Sean Bright authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Sep 11, 2013
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Rusty Newton authored
Changing text in chan_dahdi.conf sample to be accurate. (issue ASTERISK-22308) (closes issue ASTERISK-22308) Reported By: Malcolm Davenport ........ Merged revisions 398880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398881 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Russell Bryant authored
The denoise filter requires func_speex, not codec_speex. Fix this in the description of the denoise=yes option in confbridge.conf. ........ Merged revisions 398820 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398821 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Aug 30, 2013
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Rusty Newton authored
(issue ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt Jordan Review: https://reviewboard.asterisk.org/r/2811/ ........ Merged revisions 398147 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Aug 29, 2013
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Matthew Jordan authored
The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter and Tzafrir have pointed out numerous issues with the approach and have propsed an alternative in r/2757. Since it's not a time critical issue and is not worth holding up the release of 12 for it, I've gone ahead and reverted r394939 from 12/trunk and re-opened ASTERISK-21965. ........ Merged revisions 397938 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Aug 23, 2013
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Kinsey Moore authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Aug 20, 2013
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Walter Doekes authored
The autoframing option was added to chan_sip.c in r43243 (mogorman, 2006-09-19 01:32:57), but never made its way into the sample configs. Review: https://reviewboard.asterisk.org/r/2768/ ........ Merged revisions 396994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396995 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Aug 15, 2013
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Richard Mudgett authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jul 31, 2013
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Kinsey Moore authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jul 30, 2013
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Mark Michelson authored
The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*" * All files that were "res_sip*" are now "res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jul 27, 2013
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Kinsey Moore authored
This renames all files and API calls from several variants of Stasis-HTTP to ARI including: * Stasis-HTTP -> ARI * STASIS_HTTP -> ARI * stasis_http -> ari (ast_ari for global symbols, file names as well) * stasis http -> ARI Review: https://reviewboard.asterisk.org/r/2706/ (closes issue ASTERISK-22136) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jul 21, 2013
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Matthew Jordan authored
The ring tone provided in the sample indications.conf was incorrect. This patch modifies the sample ring tone to be what it should: ring = 425/400,0/200,425/400,0/2000 This brings it in line with the tone definition in DAHDI 2.7.0. (zonedata.c) (closes issue ASTERISK-21997) Reported by: Filip Jenicek patches: malaysia_ring.patch uploaded by phill (License 6277) ........ Merged revisions 394940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394941 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Matthew Jordan authored
This patch modifies the behavior of safe_asterisk in two ways: (1) It modifies the Asterisk Makefile such that safe_asterisk is always installed on a 'make install'. This was done as bugfixes in the safe_asterisk script were not applied in previous version of Asterisk without first removing the old version of the script. (2) In order to keep a newly installed version of safe_asterisk from impacting local modifications, a new config file - safe_asterisk.conf.sample - has been provided. Settings that were previously modified in safe_asterisk can be set there instead. (closes issue ASTERISK-21965) Reported by: Jeremy Kister patches: safe_asterisk.patch uploaded by jkister (License 6232) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Matthew Jordan authored
The connectedline parameter for a chan_iax2 peer was undocumented. This patch documents the options in the sample configuration file. (closes issue ASTERISK-21953) Reported by: Birger "WIMPy" Harzenetter ........ Merged revisions 394886 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394890 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jul 15, 2013
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Richard Mudgett authored
The ill conceived chan_agent is no more. It is now replaced by app_agent_pool. Agents login using the AgentLogin() application as before. The AgentLogin() application no longer does any authentication. Authentication is now the responsibility of the dialplan. (Besides, the authentication done by chan_agent did not match what the voice prompts asked for.) Sample extensions.conf [login] ; Sample agent 1001 login ; Set COLP for in between calls so the agent does not see the last caller COLP. exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>) ; Give the agent DTMF transfer and disconnect features when connected to a caller. same => n,Set(CHANNEL(dtmf-features)=TX) same => n,AgentLogin(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() [caller] ; Sample caller direct connect to agent 1001 exten => 800,1,AgentRequest(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() ; Sample caller going through a Queue to agent 1001 exten => 900,1,Queue(agent_q) same => n,Hangup() Sample queues.conf [agent_q] member => Local/800@caller,,SuperAgent,Agent:1001 Under the hood operation overview: 1) Logged in agents wait for callers in an agents holding bridge. 2) Caller requests an agent using AgentRequest() 3) A basic bridge is created, the agent is notified, and caller joins the basic bridge to wait for the agent. 4) The agent is either automatically connected to the caller or must ack the call to connect. 5) The agent is moved from the agents holding bridge to the basic bridge. 6) The agent and caller talk. 7) The connection is ended by either party. 8) The agent goes back to the agents holding bridge. To avoid some locking issues with the agent holding bridge, I needed to make some changes to the after bridge callback support. The after bridge callback is now a list of requested callbacks with the last to be added the only active callback. The after bridge callback for failed callbacks will always happen in the channel thread when the channel leaves the bridging system or is destroyed. (closes issue ASTERISK-21554) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2657/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jul 11, 2013
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David M. Lee authored
When I initially wrote the configuration support for ARI users, I determined the section type by a category prefix (i.e., [user-admin]). This is neither idiomatic Asterisk configuration, nor is it really that user friendly. This patch replaces the category prefix with a type field in the section, which is much cleaner. Review: https://reviewboard.asterisk.org/r/2664/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jul 10, 2013
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Russell Bryant authored
The SLA code within app_meetme was written before asotbj2 had been merged into Asterisk. Worse, support for reloads did not exist at first and was added later as a bolt-on feature. I knew at the time that reloading was not safe at all while SLA was in use, so the reload would be queued up to execute when the system was idle. Unfortunately, this approach was still prone to errors beyond the fact that this was the only place in Asterisk where configuration was not reloaded instantly when requested. This patch converts various SLA objects to be reference counted objects using astobj2. This allows reloads to be processed while the system is in use. The code ensures that the objects will not disappear while one of the other threads is using them. However, they will be immediately removed from the global trunk and station containers so no new calls will use them if removed from configuration. Review: https://reviewboard.asterisk.org/r/2581/ ........ Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 393929 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jul 03, 2013
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Richard Mudgett authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Richard Mudgett authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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