- Feb 06, 2014
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Rusty Newton authored
configs/pjsip.conf.sample: Configuration section naming in pjsip.conf.sample needs a little clarification There is a bit of nuance to how you name things in pjsip.conf. This is a documentation patch to at least clear it up a little for users. Review: https://reviewboard.asterisk.org/r/3180/ ........ Merged revisions 407587 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Feb 04, 2014
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Richard Mudgett authored
Thanks to Guillaume Martres for doing the necessary research to validate the change. (closes issue ASTERISK-17727) Reported by: LN Patches: use_certificate_chain.patch (license #5864) patch uploaded by st documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres ........ Merged revisions 407272 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407273 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 31, 2014
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Kevin Harwell authored
Added a "debug" configuration option for res_pjsip that when set to "yes" enables SIP messages to be logged. It is specified under the "system" type. Also added an alembic script to add the option to realtime. (closes issue ASTERISK-23038) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3148/ ........ Merged revisions 407036 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 29, 2014
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Russell Bryant authored
Closes issue ASTERISK-22662 ........ Merged revisions 406860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406861 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406862 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 27, 2014
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Walter Doekes authored
In pre-trunk versions (...12) it accepts a basic regex, which is confusing because all other regexes in asterisk are of the extended kind. Review: https://reviewboard.asterisk.org/r/3147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 21, 2014
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Walter Doekes authored
Review: https://reviewboard.asterisk.org/r/3133/ ........ Merged revisions 406079 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406080 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406081 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 17, 2014
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Rusty Newton authored
Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 16, 2014
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Kevin Harwell authored
According to the new standard for V.27 and V.32 they are able to transmit at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be updated to reflect this. Also, because of this change the default 'minrate' value was updated to be 4800. (closes issue ASTERISK-22790) Reported by: Paolo Compagnini Patches: res_fax.txt uploaded by looserouting (license 6548) ........ Merged revisions 405656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405693 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405694 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 14, 2014
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Richard Mudgett authored
The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- 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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