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${ noResults }
1554 Commits (d1e7473649afce5db9ef9e50efd56bb27c6e8f62)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
108f5fafd7 |
Improve FollowMe accept/decline DTMF string matching.
If you hit the wrong DTMF digit trying to accept/decline a FollowMe call, you had to wait for the prompt to repeat to try again. * Make FollowMe compare the last DTMF digits received to the accept/decline matching strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
|
d71d8ed995 |
Keep answered FollowMe calls until call accepted or last step times out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
|
ae1502be33 |
Add support for lightweight NAT keepalive.
If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
|
7aa0c3c64b |
Make it possible to change the minimum DTMF duration in asterisk.conf
Asterisk has a setting for the minimum allowed DTMF. If we get shorter DTMF tones, these will be changed to the minimum on the outbound call leg. (closes issue ASTERISK-19772) Review: https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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47ccc7f5d6 |
Update membermacro and membergosub documentation in queues.conf.sample.
........ Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362678 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c7cb03a975 |
Add ability to ignore layer 1 alarms for BRI PTMP lines.
Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362429 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a35c7ba8e7 |
Add option to invoke the extensions.conf stdexten using the legacy macro method.
ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in favor of the Gosub method without a means of backwards compatibility. (issue ASTERISK-18809) (closes issue ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1855/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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683eacb59a |
Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8
Prior to this patch, in order to restore that behavior, a function would have to be used on the QueueMember to make the ringinuse option do anything, which is pretty unreasonable. (closes issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1860/ ........ Merged revisions 361907 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361909 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b2f7b0c649 |
Remove a few more files related to chan_usbradio and app_rpt.
........ Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361381 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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655a8d4420 |
Introducing the log message unique call identifiers feature
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d501c2ea2d |
undoing 360785 due to merging mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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bf994f0e04 |
Introducing the log message unique call identifiers feature
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a22b56235b |
Add ability for chan_dahdi ISDN to block connected line updates per span.
Added new chan_dahdi.conf colp_send option parameter to block connected line updates per span. (closes issue ASTERISK-17025) Reported by: Michael Smith git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c369a4416b |
Massive changes in chan_unistim channel driver. Include many fixes in channel driver operation and add additional functionality:
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14 years ago |
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f5fda0eb74 |
Transition app_page to using app_confbridge internally for the conference bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles.
Review: https://reviewboard.asterisk.org/r/1754/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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706f79d122 |
Handle numeric columns for eventtype properly in cel_odbc
Patch also implements correct handling of datetime2 and datetimeoffset new datatypes in SQL Server 2008 and 2008 R2. (closes issue ASTERISK-17548) Review: https://reviewboard.asterisk.org/r/1160/ Review: https://reviewboard.asterisk.org/r/1804/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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deded253b3 |
Set snarkiness = 0 in cdr_adaptive_odbc.conf.sample
........ Merged revisions 358438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358441 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358444 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a0f8821749 |
Add dialtone_detect option for analog incoming calls.
For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered. If dialtone is
detected, the call is hung up.
no: Disabled. (Default)
yes: Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always: Look for dialtone for the entire call. Dialtone may return
if the far end hangs up first.
dialtone_detect=yes
dialtone_detect=5000
dialtone_detect=always
(closes issue ASTERISK-19316)
Reported by: Jeremy Pepper
Patch by: Jeremy Pepper
Tested by: rmudgett,Jeremy Pepper
Review: https://reviewboard.asterisk.org/r/1737/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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1d9203eeef |
Tab to spaces and text change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2d92c50ce3 |
Beef up the IAX2 sample configuration a bit and fix some formatting issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0b988da21c |
Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE
(issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) ........ Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357497 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1fac2fba4b |
Deprecated macro usage for connected line, redirecting, and CCSS
This commit adds GoSub alternatives to connected line, redirecting, and CCSS macro hooks so that macro can finally be deprecated. This also adds deprecation warnings for those features when used and in documentation. Review: https://reviewboard.asterisk.org/r/1760/ (closes issue SWP-4256) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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299dd5d4fc |
Adds an option to sip.conf that prevents diversion headers from being added.
send_diversion=no will prevent Diversion headers from being added to SIP requests. This doesn't prevent Diversion from being added with dialplan such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3816fdde94 |
Don't allow trunkfreq to be greater than 1000ms.
........ Merged revisions 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355794 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355795 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a78b0af5ea |
Re-commit the verbose branch.
This change permits each verbose destination (consoles, logger) to have its own concept of what the verbosity level is. The big feature here is that the logger will now be able to capture a particular verbosity level without condemning each console to need to suffer that level of verbosity. Additionally, a stray 'core set verbose' will no longer change what will go to the log. Review: https://reviewboard.asterisk.org/r/1599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1cdff5b720 |
Don't enable sqlite3 CDRs by default in sample configs.
........ Merged revisions 355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355229 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a955a4770f |
Fix reconnecting to pgsql database after connection loss.
There can only be one database connection in res_config_pgsql just like res_config_sqlite. If the connection is lost, the connection may not get reestablished to the same database if the res_pgsql.conf and extconfig.conf files are inconsistent. * Made only use the configured database from res_pgsql.conf. * Fixed potential buffer overwrite of last[] in config_pgsql(). (closes issue ASTERISK-16982) Reported by: german aracil boned Review: https://reviewboard.asterisk.org/r/1731/ ........ Merged revisions 354953 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354959 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354964 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e5c51ee44c |
Add auto_force_rport and auto_comedia NAT options
This patch adds the auto_force_rport and auto_comedia NAT options. It also converts the nat= setting to a list of comma-separated combinable options: no, force_rport, comedia, auto_force_rport, and auto_comedia. nat=yes remains as an undocumented option equal to "force_rport,comedia". The first instance of 'yes' or 'no' in the list stops parsing and overrides any previously set options. If an auto_* option is specified with its non-auto_ counterpart, the auto setting takes precedence. This patch builds upon the patch posted to ASTERISK-17860 by JIRA user pedro-garcia. (closes issue ASTERISK-17860) Review: https://reviewboard.asterisk.org/r/1698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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055a19e128 |
Replace res_ais with a new module, res_corosync.
This patch removes res_ais and introduces a new module, res_corosync. The OpenAIS project is deprecated and is now just a wrapper around Corosync. This module provides the same functionality using the same core infrastructure, but without the use of the deprecated components. Technically res_ais could have been used with an AIS implementation other than OpenAIS, but that is the only one I know of that was ever used. Review: https://reviewboard.asterisk.org/r/1700/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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71a8457d53 |
Support schema selection in cdr_adaptive_odbc
Asterisk now supports using ODBC with databases where a single schema must be selected. Previously, INSERTs would fail because they did not take into account extra fields cause by having multiple schemas. This also corrects some SQL resource leaks. (closes issue ASTERISK-17106) Patch-by: Alexander Frolkin Patch-by: Tilgnman Lesher git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353964 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0f4489dc0f |
Fix TLS port binding behavior as well as reload behavior:
* Removes references to tlsbindport from http.conf.sample and manager.conf.sample * Properly bind to port specified in tlsbindaddr, using the default port if specified. * On a reload, properly close socket if the service has been disabled. A note has been added to UPGRADE.txt to indicate how ports must be set for TLS. (closes issue ASTERISK-16959) reported by Olaf Holthausen (closes issue ASTERISK-19201) reported by Chris Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas Review: https://reviewboard.asterisk.org/r/1709 ........ Merged revisions 353770 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353820 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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797d633139 |
Remove inconsistency in CEL eventtype for user defined events.
The CEL eventtype field for ODBC and PGSQL backends should be USER_DEFINED instead of the user defined event name supplied by the CELGenUserEvent application. If the field is output as a number, the user defined name does not have a value and is always output as 21 for USER_DEFINED and the userdeftype field would be required to supply the user defined name. The following CEL backends (cel_odbc, cel_pgsql, cel_custom, cel_manager, and cel_sqlite3_custom) can be independently configured to remove this inconsistency. * Allows cel_manager, cel_custom, and cel_sqlite3_custom to behave the same way. (closes issue ASTERISK-17189) Reported by: Bryant Zimmerman Review: https://reviewboard.asterisk.org/r/1669/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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973aeabf2d |
Redocuments sip types peer, user, friend in sip.conf.sample
There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique ........ Merged revisions 352511 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352512 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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de09749470 |
Add an announcement option to music-on-hold - plays sound when put on hold/between songs
This is a feature patch which allows an 'announcement' option to be specified in musiconhold.conf which should be set to the name of a sound. If a valid sound is specified for this option, then it will be played on that music on hold class whenever a channel bound to that class is put on hold as well as when Asterisk is able to detect that a song has ended before starting the next song (excludes external players). (closes ASTERISK-18977) Reported by: Timo Teräs Patches: asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352134 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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778fa4abaf |
Various parking improvements.
* Adds per-parking lot options comebackcontext and comebackdialtime * Makes comebacktoorigin settable per parking lot * Sets a PARKER channel variable when comebacktoorigin is disabled (closes issue ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches: asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231 with updates by me. Review: https://reviewboard.asterisk.org/r/1674 Review: https://reviewboard.asterisk.org/r/963 Reviewed by Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ee4cf38a27 |
Adds pjmedia probation concepts to res_rtp_asterisk's learning mode.
In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ ........ Merged revisions 351287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351289 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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835067a526 |
Correct eventtype names in cel_odbc and cel_pgsql sample files
........ Merged revisions 350733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350734 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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523c95e146 |
Add missing CEL logging fields to various CEL backends.
Multiple revisions 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51 -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging fields to various CEL backends. * Add missing eventextra to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and EventExtra to cel_manager.c. * Add missing userdeftype support for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 Jan 2012) | 8 lines Use compatible names for event extra data for various CEL backends. * Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190) ........ Merged revisions 350555,350571 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350585 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350605 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b0243fb57c |
Allow overriding of IMAP server settings on a user by user basis
This patch allows the imapserver, imapport, and imapflags settings to be overridden for any voicemail user. It also documents the settings in the sample voicemail.conf file, and updates the voicemail schema to allow storage of those columns. (closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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19a4928fee |
INFO/Record request configurable to use dynamic features
Adds two new options to SIP peers allowing them to specify features (dynamic or builtin) to use when sending INFO/record requests. Recordonfeature activates whatever feature is specified when recieving a record: on request while recordofffeature activates whatever feature is specified when receiving a record: off request. Both of these features can be disabled by setting the feature to an empty string. (closes issue ASTERISK-16507) Reported by: Jon Bright Review: https://reviewboard.asterisk.org/r/1634/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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03596bcb47 |
chan_sip autocreatepeer=persist option for auto-created peers to survive reload
This patch moves destruction of sip peers to immediately after the general section of sip.conf is read so that autocreatepeer setting can be read before deletion of peers. If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting will be skipped when purging the current SIP peer list. (closes ASTERISK-16508) Reported by: Kirill Katsnelson Patches: 017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d30a7ba3ce |
Correct two flaws in sip.conf.sample related to AST-2011-013.
* The sample file listed *two* values for the 'nat' option as being the default. Only 'force_rport' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. ........ Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348517 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348518 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1b0741c7db |
Voicemail with the saycid option will now play a caller's name based on cid if available.
In order to check the availability of the caller's name, app_voicemail will check for an audio file in <astspooldir>/recordings/callerids/ This change sets a precedent for where to put recordings of names. Currently the idea is that recordings here could also be used for applications like confbridge and meetme to find recorded names in this folder from callerid (when another recording isn't available) (closes issue ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by Russel Brown (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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480d46f92c |
Add and document PARKEDCALL variable set during timeout
PARKEDCALL variable tracks which parking lot the call was last parked in. This can be used afterwards for flow control when returntoorigin is set to off. I went ahead and documented both this and the existing variable set during timeout (PARKINGSLOT) in the sample features.conf since there was no prior mention of variables being set during timeout. (closes issue ASTERISK-16239) Reported By: Clod Patry Patches: M17503.diff uploaded by Clod Patry (license 5138) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c3f703330b |
Fix accidental use of tabs instead of spaces from previous features.conf.sample change
........ Merged revisions 348157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348158 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2d0491d432 |
Document PARKINGSLOT variable in features.conf.sample
(issue ASTERISK-16239) ........ Merged revisions 348154 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348155 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348156 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9057aa20b6 |
Backed out core changes from r346391
During testing, it was discovered that there were a number of side effects introduced by r346391 and subsequent check-ins related to it (r346429, r346617, and r346655). This included the /main/stdtime/ test 'hanging', as well as the remote console option failing to receive the appropriate output after a period of time. I only backed out the changes to main/ and utils/, as this was adequate to reverse the behavior experienced. (issue ASTERISK-18974) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e2597678b1 |
Update sample configs to put incoming calls into context public.
* Add warning about the SIP allowguest option in context public. (closes issue ASTERISK-14122) Reported by: Alec Davis Review: https://reviewboard.asterisk.org/r/719/ ........ Merged revisions 347953 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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980ab2d018 |
Add ASTSBINDIR to the list of configurable paths
This patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH. (closes issue ASTERISK-18959) Review: https://reviewboard.asterisk.org/r/1613/ ........ Merged revisions 347344 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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83cd844b82 |
Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.
The STUN socket must remain open between polls or the external address seen by the STUN server is likely to change. However, if the STUN request poll fails then the STUN server address needs to be re-resolved and the STUN socket needs to be closed and reopened. * Re-resolve the STUN server address and create a new socket if the STUN request poll fails. * Fix ast_stun_request() return value consistency. * Fix ast_stun_request() to check the received packet for expected message type and transaction ID. * Fix ast_stun_request() to read packets until timeout or an associated response packet is found. The stun_purge_socket() hack is no longer required. * Reduce ast_stun_request() error messages to debug output. * No longer pass in the destination address to ast_stun_request() if the socket is already bound or connected to the destination. (closes issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1595/ ........ Merged revisions 346700 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346701 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |