mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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1.6.0.8
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10.7.0-digiumphones
10.7.0-digiumphones-rc1
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10.9.0
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certified/1.8.11-cert1
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${ noResults }
22376 Commits (745fcdbffe207d1cbbfabac3c8e8ea1a33df6a48)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Richard Mudgett
|
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 |
13 years ago |
Joshua Colp
|
fd100261f3 |
Don't try to play sound files that do not exist.
(closes issue ASTERISK-19188) Reported by: slesru ........ Merged revisions 354938 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354939 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Jason Parker
|
6749b6e2be |
Fix a voicemail memory leak with heard/deleted messages.
open_mailbox() was changed quite a long time ago to allocate this memory. close_mailbox() should have been changed to be responsible for freeing it. ........ Merged revisions 354889 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354890 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
8af49f158a |
Fix AMI Redirect ExtraChannel not redirecting to the same exten and context.
The astman_get_header() never returns NULL so the check by the code for
NULL would never fail.
(closes issue ASTERISK-16974)
Reported by: Nuno Borges
Patches:
0018325.patch (license #6116) patch uploaded by Nuno Borges (modified)
........
Merged revisions 354835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 354836 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
Matthew Jordan
|
a8cf4dc2b5 |
Fix IMAP app_voicemail compilation issue introduced in r354429
This simply fixes the compilation issue introduced in r354429 by re-adding the 'quote' variable. (closes issue ASTERISK-19337) Reported by: John Taylor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354799 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Terry Wilson
|
8100a1703d |
Note that CDRs are immutable once a bridge is torn down
CDRs cannot be modified after a bridge is torn down, (e.g. after Dial() returns) even though the CDR() function may be called. Since modifying the CDR code to change this behavior could very easily break all kinds of things, this patch just documents this limitation. (closes issues ASTERISK-16923) Review: https://reviewboard.asterisk.org/r/1720/ ........ Merged revisions 354749 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354750 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Kinsey Moore
|
6225c6cadc |
Fix parsing of SIP headers where compact and non-compact headers are mixed
Change parsing of SIP headers so that compactness of the header no longer influences which header will be chosen. Previously, a non-compact header would be chosen instead of a preceeding compact-form header. (closes issue ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/ ........ Merged revisions 354702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354703 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
25e344168e |
Make the config parser remove escaping backslashes
The config parser in Asterisk does not currently remove a backslash that is used to escape a semicolon which would otherwise be interpreted as the start of a comment. The change here causes that backslash to be removed, but does not create a real escape system in the config parser. The biggest complication with a real escape system would be breaking existing configs everywhere (parsing \\ as \ and breaking on escaped non-semicolon characters) even though it would be the "right" way to do things. (closes issue ASTERISK-17121) Review: https://reviewboard.asterisk.org/r/1724/ ........ Merged revisions 354655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354656 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
|
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 |
13 years ago |
Mark Michelson
|
8f5c33f95a |
Adding reload support to res_fax.so
(closes issue ASTERISK-16712) reported by Frank DiGennaro Review: https://reviewboard.asterisk.org/r/1713 ........ Merged revisions 354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354546 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
dff9b61f5c |
Clean-up of minor formatting issues in r354542/3/4
rmudgett pointed out some formatting issues in the check-in for ASTERISK-19290. This cleans those up. Review: https://reviewboards.asterisk.org/r/1722/ ........ Merged revisions 354547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354548 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354549 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
ba08e9f4d6 |
Fix SIP INFO DTMF handling for non-numeric codes
In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared again '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........ Merged revisions 354542 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354543 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
16fbc7e902 |
Fix some compile problems from the 'cppcheck' patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
5816c60654 |
Fix crash in ParkAndAnnounce.
Well, thats embarrasing. I forgot to initialize the caller_id storage. (closes issue ASTERISK-19311) Reported by: tootai Tested by: rmudgett ........ Merged revisions 354495 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354496 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Russell Bryant
|
747cd61edf |
Remove some unnecessary locking from ast_hangup().
This patch removes some unnecessary locking of the channels container in ast_hangup(). The reason this came up is that this lock can very quickly block the entire system. If any of the channel cleanup code decides to block, it causes a problem for the whole system. For example, when audiohooks get destroyed, if that blocks for a while waiting on the mixmonitor thread to exit because it's busy blocking on some I/O, it causes a problem for many other threads in the meantime. Review: https://reviewboard.asterisk.org/r/1712/ ........ Merged revisions 354492 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354493 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354494 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Kevin P. Fleming
|
2ce189c5b8 |
Revision 354046 added res_corosync as a replacement for res_ais, but didn't
actually remove res_ais. This commit removes it. In addition, the 'install_prereq' script has been updated to no longer install AIS dependency packages, and instead install Corosync packages instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354459 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Terry Wilson
|
3342183016 |
Add callbackextension matching & realtime callbackextensions
This patch is based on the one by David Vossel, developer extrodinaire, at https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the same host/port, but differing callbackextensions, it chooses the peer with the matching callbackextension. Since callbackextension creates an outbound registration with the callbackextension as the Contact address, matching an incoming request by that (in addition to the host/port) makes a lot of sense. This patch also adds support for callbackextension to realtime by querying all peers with callbackextensions on reload and adding registrations for them. (closes issue ASTERISK-13456) Review: https://reviewboard.asterisk.org/r/344/ Review: https://reviewboard.asterisk.org/r/1717/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Kevin P. Fleming
|
f0e321b88a |
Restore some variables removed by the 'cppcheck' patch that were actually needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Walter Doekes
|
db24fc2523 |
Avoid cppcheck warnings; removing unused vars and a bit of cleanup.
Patch by: Clod Patry Review: https://reviewboard.asterisk.org/r/1651 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Kinsey Moore
|
0adeb88318 |
Add CHANGES documentation for the "pri set debug" bitmask change
(related to ASTERISK-17159) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354395 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Terry Wilson
|
8ba2d70602 |
Fix multiple SIP realtime issues
1. Set lastms to 0 when clearing instead of "" 2. Don't set ipaddr or port to the string "(null)" when they are empty 3. Add missing required fields, set default for lastms to 0, and modify the length of the ipaddr field to 45 in the Postgresql realtime.sql file. (closes issue ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/ ........ Merged revisions 354348 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354349 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354360 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Sean Bright
|
994d4d019c |
Continuation of last patch - since LIVE_AST_LD_PATH_EXTRA will now never
be empty, don't check for it, instead of check if LD_LIBRARY_PATH is already set and if so, append LIVE_AST_LD_PATH_EXTRA properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Sean Bright
|
8e79e31aa5 |
Include live/usr/lib in the shared library search path to that we pick up
libasteriskssl.so at run time when using live_ast. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354313 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Sean Bright
|
3fda975b9d |
Whitespace only (remove trailing spaces)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354312 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jonathan Rose
|
b08d91c3dc |
Fix column duplication bug in module reload for cdr_pgsql.
Prior to this patch, attempts to reload cdr_pgsql.so would cause the column list to keep its current data and then add a second copy during the reload. This would cause attempts to log the CDR to the database to fail. This patch also cleans up some unnecessary null checks for ast_free and deals with a few potential locking problems. (closes issue ASTERISK-19216) Reported by: Jacek Konieczny Review: https://reviewboard.asterisk.org/r/1711/ ........ Merged revisions 354263 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354270 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354275 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
84642c728f |
Improved documentation of CLI "dialplan add extension" command.
* Documented dialplan add extension <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance of command without the app-data value. There are many applications that do no need any parameters so it is silly to require that field for all commands. * Fixed a couple ast_malloc/ast_free mismatches with ast_add_extension2() calls. (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested by: rmudgett ........ Merged revisions 354216 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354217 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
a4f5d2c2ef |
Restore alternate SIG_PRI_DEBUG_DEFAULT meaning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Kinsey Moore
|
49ed50d8ac |
Allow more control over the output of pri debug
This changes the debuglevel of 'pri set debug' to a bit mask allowing the user to independently select bits of output: 1 libpri internals including state machine 2 Decoded Q.931 messages 4 Decoded Q.921 headers 8 raw hex dump of the full frames Additionally, this ensures that the meaning of "on" does not change and intrudces intense and hex to simplify usage. (closes issue ASTERISK-17159) Original-patch-by: wimpy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
d162e85978 |
Add missing headers to AMI UnParkedCall event to uniquely identify the call.
The AMI UnParkedCall event was missing the Parkinglot and Uniqueid headers that the AMI ParkedCall event contains. (closes issue ASTERISK-19240) Reported by: Michael Yara ........ Merged revisions 354116 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354119 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Joshua Colp
|
afdd96712c |
Make the 'c' option to MeetMe work even if the 'q' option is used.
(closes issue ASTERISK-17053) Reported by: justdave git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354084 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Russell Bryant
|
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 |
|
Jonathan Rose
|
a898eb4d07 |
Fixes deadlocks occuring in chan_agent due to r335976
Bad locking order was added to chan_agent to prevent segfaults from having no locking in a patch by irroot. This patch addresses the bad locking order by releasing locks before getting the right locking order to stop deadlocks from occuring when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1708/ ........ Merged revisions 353999 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354000 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354001 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Kinsey Moore
|
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 |
|
Jonathan Rose
|
79979313e8 |
Fixes a segfault occuring when performing attended transfer with FAXOPT(gateway)=yes
(closes issue ASTERISK-19184) Reported by: Alexandr ........ Merged revisions 353962 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353963 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Kinsey Moore
|
29318afc15 |
Ensure entering T.38 passthrough does not cause an infinite loop
After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is shut down and removed. If the descriptor happened to have data ready when the removal occured then Asterisk would go into an infinite loop trying to read data that it can never actually access. This change disables the audio RTCP file descriptor for the duration of the T.38 transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan Vrban ........ Merged revisions 353915 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353916 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
63c5eaee43 |
Restore the 'w' modifier support for ISDN spans. Dial(DAHDI/g0/1234w888)
This feature also causes the sending complete ie to be sent for switch types that do not automatically send the ie. (EuroISDN/ETSI) The main difference between dialing Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the sending of the sending complete ie. (closes issue ASTERISK-19176) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 353867 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353868 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353872 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Mark Michelson
|
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 |
|
Jonathan Rose
|
5164196972 |
Fix sip show peers port output, align columns, and fix ami port output.
A previous patch I committed from ASTERISK-16930 unexpectedly changed some output for the AMI action "sippeers" which this patch changes back. Also, this aligns the output for the cli command "sip show peers" and fixes another issue that patch introduced by using ast_sockaddr_stringify calls multiple times without immediately using the pointer. I also went ahead and did a little janitorial work to clean up whitespace in _sip_show_peers. (issue ASTERISK-16930) (closes issue ASTERISK-19281) Reported by: Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by Walter Doekes (license 5674) ........ Merged revisions 353769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353771 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Jonathan Rose
|
0e334d427b |
Use ast_sockaddr_stringify_fmt wrappers for various functions in chan_sip
There are a number of cleaner looking wrappers for ast_sockaddr_stringify_fmt available which are slightly more readable than using a direct call to ast_sockaddr_stringify_fmt. This patch switches a number of those calls in chan_sip to use those wrappers and is generally harmless. (Closes issue ASTERISK-16930) Reported by: Michael L. Young Patches: chan_sip-broken-registration-1.8.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 353720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353721 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353725 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
23bc964e1c |
Constify some more channel driver technology callback parameters.
Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
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 |
|
Richard Mudgett
|
a99b3c817b |
Fix ExtenSpy and simplify the channel search functions.
When ast_channel name was opaquified, the channel search functions did not get converted correctly. As a result ExtenSpy which uses a channel iterator search by exten@context could never find anything. * Updated the doxygen documentation for the search functions in channel.h. Review: https://reviewboard.asterisk.org/r/1702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Sean Bright
|
544333b435 |
Resolve an overlap in the ast_audiohook_flags values.
AST_AUDIOHOOK_TRIGGER_WRITE and AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused unintended side effects. This patch moves AST_AUDIOHOOK_TRIGGER_WRITE, and updates AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention. This will affect existing modules that use these flags, so be sure to recompile as necessary. (closes issue ASTERISK-19246) Reported by: feyfre ........ Merged revisions 353598 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353599 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Matthew Jordan
|
863493118b |
Added clarification for the VERBOSITY setting to etc_default_asterisk
Clarified that using the VERBOSITY setting in etc_default_asterisk is the same as using the -v command line switch, which causes Asterisk to launch in console mode. (closes issue ASTERISK-17030) Reported by: Jonas ........ Merged revisions 353550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353551 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Terry Wilson
|
5861bab06d |
Allow res_calendar to be unloaded
The calendaring tech modules depend on res_calendar and initially res_calendar just bumped the use count so that it couldn't be unloaded. res_calendar can potentially create many threads and I've seen issues where the Asterisk shutdown has failed where it looked like these threads could be the culprit. This patch adds unload support for res_calendar. Unloading res_calendar will also unload the dependant tech modules as well. (closes issue ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/ ........ Merged revisions 353502 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353503 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353504 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
2d7a40de58 |
Fix memory leak in error paths for action_originate().
* Fix memory leak of vars in error paths for action_originate(). * Moved struct fast_originate_helper tech and data members to stringfields. * Simplified ActionID header handling for fast_originate(). * Added doxygen note to ast_request() and ast_call() and the associated channel callbacks that the data/addr parameters should be treated as const char *. Review: https://reviewboard.asterisk.org/r/1690/ ........ Merged revisions 353454 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353463 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Terry Wilson
|
de57235ac6 |
Re-link peers by IP when dnsmgr changes the IP
Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. This patch adds a function to the dnsmgr API that calls an update callback instead of blindly updating the address itself. It also moves calls to ast_dnsmgr_release outside of the destructor functions and into cleanup functions that are called when we no longer need the objects and increments the refcount of the objects using dnsmgr since those objects are stored on the ast_dnsmgr_entry struct. A helper function for returning the proper default SIP port (non-tls vs tls) is also added and used. This patch also incorporates changes from a patch posted by Timo Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/ ........ Merged revisions 353371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353397 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353418 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Alec L Davis
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f92d6412ab |
Merged revisions 353369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353369 | alecdavis | 2012-01-31 11:42:28 +1300 (Tue, 31 Jan 2012) | 9 lines Merged revisions 353368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq numbers. Missed in R353320 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Alec L Davis
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0ccc1f5274 |
Merged revisions 353321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353321 | alecdavis | 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) | 25 lines Merged revisions 353320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kevin P. Fleming
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c6489d7b32 |
Correct serious flaw in the top-level Makefile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353319 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |