mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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10.8.0
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certified/1.8.11-cert1
certified/1.8.11-cert10
certified/1.8.11-cert2
certified/1.8.11-cert3-rc1
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certified/1.8.11-cert5-rc2
certified/1.8.11-cert6
certified/1.8.11-cert7
certified/1.8.11-cert8
certified/1.8.11-cert9
certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
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certified/1.8.28-cert1
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certified/1.8.28-cert2
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certified/1.8.6-cert1
certified/11.2-cert1
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certified/11.6-cert1
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certified/11.6-cert10
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certified/11.6-cert5
certified/11.6-cert6
certified/11.6-cert7
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certified/13.1-cert1
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${ noResults }
22224 Commits (ebe2c33b7239b2881007efa3efd8d8a859f0b653)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
463bfdb400 |
Fix regression when using tcpenable=no and tlsenable=yes.
The tlsenable settings are tucked away in main/tcptls.c, so I missed them when resolving ASTERISK-18837. This should resolve the test suite breakage of the sip tls tests. Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt Jordan ........ Merged revisions 347718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347727 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
e1884139c4 |
Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>'
r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when attempting to set penalty on an interface without specifying a queue in the queue set penalty CLI command. In addition, no attempt would be made whatsoever to perform the penalty setting on all the queues in the core list with either the cli command or the non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to document and rename some functions required by this command to better represent what they actually do. Oh yeah, and the use of this command without specifying a specific queue actually works now. Review: https://reviewboard.asterisk.org/r/1609/ ........ Merged revisions 347656 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347658 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
3f13a41886 |
Mark channel running the h exten with the soft-hangup flag.
When a bridge is broken, ast_bridge_call() might execute the h exten on
the calling channel. However, that channel may not have been the channel
that broke the bridge by hanging up. The channel executing the h exten
must be in a hung up state so things like AGI run in the correct mode.
* Make sure ast_bridge_call() marks the channel it is executing the h
exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to
match the pbx.c main dialplan execution loop when it executes the h
exten.)
(closes issue ASTERISK-18811)
Reported by: David Hajek
Patches:
jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: David Hajek, rmudgett
........
Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347600 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
|
|
e279b30f5a |
Don't crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. (closes issue ASTERISK-18805) ........ Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347532 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
5952117559 |
Fix segfault on answer.
Fix a segfault if an attempt to answer a call is made between when the inbound call gives up (and the channel is removed) and when the device is notified and removes the call from the device. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
395814c33e |
Update AMI Getvar and Setvar documentation about supplying a channel name.
(closes issue ASTERISK-18958)
Reported by: Red
Patches:
jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett
........
Merged revisions 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347439 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
|
|
8e94432d9a |
Fix: Meetme recording variables from realtime DB use null entries over channel variables
Meetme would attempt to substitute the realtime values of RECORDING_FILE and RECORDING_FORMAT from the meetme db entry instead of using the channel variable set for those variables in spite of those database entries being NULL or even lacking a column to represent them. (closes issue ASTERISK-18873) Reported by: Byron Clark Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157) ........ Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347383 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347395 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
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 |
15 years ago |
|
|
7e634c21f8 |
Make SIP INFO messages for dtmf-relay signals case insensitive.
(closes issue ASTERISK-18924) Reported by: Kevin Taylor ........ Merged revisions 347292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347293 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347294 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
9b33408ba1 |
Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten
If waitExten specifies a music class to use with its music on hold option, it will use CHANNEL(musicclass) instead if that channel variable has been set on the initiating channel. This documents that behavior in the waitExten app so that this can be known without checking the documentation of the code in function local_ast_moh_start. (closes issue ASTERISK-18804) ........ Merged revisions 347239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347240 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fd64bb66f9 |
Add VM_INFO() dialplan function to gather information about a mailbox.
Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname, language, locale, pager, password, tz. (closes issue ASTERISK-18634) Patch by: Kris Shaw Review: https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e14f572132 |
Don't allow transport=tcp when tcpenable=no.
When tcpenable=no, sending to transport=tcp hosts was still allowed. Resolving the source address wasn't possible and yielded the string "(null)" in SIP messages. Fixed that and a couple of not-so-correct log messages. (closes issue ASTERISK-18837) Reported by: Andreas Topp Review: https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan ........ Merged revisions 347166 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347167 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7bdaa31d25 |
Add regression tests for issue ASTERISK-18838.
Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan ........ Merged revisions 347131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347146 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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03fd2c0c94 |
The voicemail [general] zonetag and locale variables weren't loaded
until after the mailboxes were initialized. This caused the settings to be unset for those mailboxes until a reload was performed. (closes issue ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570 Reviewed by: Matt Jordan ........ Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347124 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347157 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ca41b4aba0 |
Doubly linked lists unit test and update to implementation.
Update the doubly linked list implementation. Now safe traversing can insert before and after the current node when traversing in either direction. Updated the linked lists unit test test_linkedlist to also test doubly linked lists. The old test_dlinkedlist requires a manual check of results and probably should be removed. Review: https://reviewboard.asterisk.org/r/1569/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347110 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b8dba87a4c |
Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Reported by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches: blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283) Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged revisions 347058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347068 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347069 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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938b642245 |
Restore call progress code for analog ports.
Extracting sig_analog from chan_dahdi lost call progress detection
functionality.
* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
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Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 347007 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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c5fe1cfdc0 |
Resolve duplicate label used in multiple priorities for the same extension.
Prior to this patch, if labels with the same name were used for different priorities in the same extension, the new label would be accepted, but it would be unusable since attempts to reach that label would just go to the first one. Now pbx.c detects this, generates a warning in logs, and culls the label before adding it to the dialplan. (closes issue ASTERISK-18807) Reported by: Kenneth Shumard Patches: pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........ Merged revisions 346954 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346955 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ae61df53f1 |
Fix chan_jingle/gtalk load regression introduced in r346087
Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland ........ Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346952 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346953 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6209c5a894 |
For SIP REGISTER fix domain-only URIs and domain ACL bypass.
The code that allowed admins to create users with domain-only uri's had stopped to work in 1.8 because of the reqresp parser rewrites. This is fixed now: if you have a [mydomain.com] sip user, you can register with useraddr sip:mydomain.com. Note that in that case -- if you're using domain ACLs (a configured domain list) -- mydomain.com must be in the allow list as well. Reviewboard r1606 shows a list of registration combinations and which SIP response codes are returned. Review: https://reviewboard.asterisk.org/r/1533/ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes issue ASTERISK-18741) ........ Merged revisions 346899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346900 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fa4a7dcc45 |
Update SIP MESSAGE To parsing to correctly handle URI
The previous patch (r346040) incorrectly parsed the URI in the presence of a port, e.g., user@hostname:port would fail as the port would be double appended to the SIP message. This patch uses the parse_uri function to correctly parse the URI into its username and hostname parts, and places them in the correct fields in the sip_pvt structure. (issue ASTERISK-18903) Review: https://reviewboard.asterisk.org/r/1597/ ........ Merged revisions 346856 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fa116b5e68 |
implement nat option for rtp channels with ooh323
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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db0ed2e5c8 |
Merged revisions 346763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines Merged revisions 346762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines process null frame pointer returned by ast_rtp_instance_read correctly (closes issue ASTERISK-16697) Reported by: under Patches: segfault.diff (License #5871) patch uploaded by under ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 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 |
15 years ago |
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39424ebad2 |
Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing.
183 is actually a session progress message. (closes issue ASTERISK-18925) Reported by: Sebastian Denz Tested by: jrose Patches: asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139) ........ Merged revisions 346697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346698 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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56b21b4683 |
Remove the few places where we try to ast_verbose() without a newline.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3106f64eac |
Fix edge case for overflow buffer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9ef171ffe0 |
r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines
Cleaning up chan_sip/tcptls file descriptor closing. This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged revisions 346564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346565 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fb4c483eb7 |
Reverting 346525 due to accidental patch against trunk instead of 1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346563 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6fa827b5d0 |
Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cb21847e03 |
Update queues.conf.sample documentation.
Update the documentation surrounding the use of MONITOR_EXEC to make it more clear
that it can be used for both Monitor() and MixMonitor() usage.
(closes issue ASTERISK-17413)
Reported by: David Woolley
Patches:
issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026)
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Merged revisions 346472 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 346473 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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b63c61953b |
Fix compilation of utilities (caught by Bamboo).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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77b670c4ab |
Allow each logging destination and console to have its own notion of the verbosity level.
Review: https://reviewboard.asterisk.org/r/1599 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d7dec4f14f |
Merged revisions 346349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines Fixes memory leak in message API. The ast_msg_get_var function did not properly decrement the ref count of the var it retrieves. The way this is implemented is a bit tricky, as we must decrement the var and then return the var's value. As long as the documentation for the function is followed, this will not result in a dangling pointer as the ast_msg structure owns its own reference to the var while it exists in the var container. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346350 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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edaf970c38 |
Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Fra Review: https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter Doekes ........ Merged revisions 346292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346293 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346294 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7d9ba4875b |
Fix calls to ast_get_ip() not initializing the address family.
........ Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346240 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2ee874178e |
Minor cleanup in chan_sip get_msg_text() function.
In r116240, get_msg_text() got an extra parameter to fix the unwanted addition of trailing newlines to SIP MESSAGE bodies. This caused all linefeeds to be trimmed, which isn't right either. This is a stop-gap; the right fix is to return the original SIP request body. Review: https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan ........ Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346198 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b7aee9ebc9 |
Fix ast_str_truncate signedness warning and documentation.
Review: https://reviewboard.asterisk.org/r/1594 ........ Merged revisions 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346145 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e6ca768081 |
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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76394a727f |
Fixed SendMessage stripping extension from To: header in SIP MESSAGE
When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ ........ Merged revisions 346040 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6d05a31d9f |
Resume playing existing hold music for cached realtime MOH
As a result of the fix for ASTERISK-18039, realtime caching MOH no longer properly resumes playing back a file between different holds in the same call. This is because scanning for new files causes the existing file array to be emptied and we were just comparing that the saved pointer to the filename matched the pointer to the filename in a particular position in the array. An easy fix is to save the filename instead of a pointer to it and then do a strcmp instead of comparing the addresses. (closes issue ASTERISK-18912) Review: https://reviewboard.asterisk.org/r/1596/ ........ Merged revisions 346030 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346031 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346033 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f59322f724 |
Added support level for new modules
........ Merged revisions 346029 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2b3e28f88c |
Fix dnsmgr entries to ask for the same address family each time.
The dnsmgr refresh would always get the first address found regardless of the original address family requested. So if you asked for only IPv4 addresses originally, you might get an IPv6 address on refresh. * Saved the original address family requested by ast_dnsmgr_lookup() to be used when the address is refreshed. ........ Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345977 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d777e15792 |
Clarify why the AST_LOG_* macros exist next to the LOG_* macros.
(issue ASTERISK-17973) ........ Merged revisions 345923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345924 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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51ce2669af |
Add missing sound_only_one config variable
(closes issue ASTERISK-18895)
Reported by: zvision
Patches:
conf_config_parser.diff (license #5755) patch uploaded by zvision
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Merged revisions 345882 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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32d0faac9c |
Default to nat=yes; warn when nat in general and peer differ
It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345830 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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298d015828 |
Add #tryinclude statement
This provides the same functionality as #include however an asterisk module will still load if the filename does not exist. Review: https://reviewboard.asterisk.org/r/1476/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6e7359f594 |
Update the documentation to better clarify how the existing commands work.
Review: https://reviewboard.asterisk.org/r/1593/ ........ Merged revisions 345682 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345683 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6b8b13ec2d |
Fix a change in behavior in 'database show' from 1.8.
In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key. Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev ........ Merged revisions 345640 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cd9680e241 |
Accidentally readded sipfriends.sql in r345560. This was removed
in r342871 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345601 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |