mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
22408 Commits (70c5ac6635062efe03b51fb6c7afd30c22028be9)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Sean Bright
|
1c971ae604 |
Make 'iax2 show callnumber usage' output make sense when an IP is passed in.
........ Merged revisions 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356108 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Kinsey Moore
|
4585ec1bbf |
Add missing newline to ccss state change notification
Move along, nothing to see here... ........ Merged revisions 356074 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356075 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Terry Wilson
|
57f42bd74f |
ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
25e5eb3b96 |
Remove spurious warning when 'qualifyfreqnotok' is set successfully.
(closes issue ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512) ........ Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355998 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355999 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
db487bd7f8 |
This was a LOG_NOTICE, so roll it back.
........ Merged revisions 355952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355953 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
2bd6649a93 |
Change some debug messages from LOG_DEBUG to ast_debug.
........ Merged revisions 355949 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355950 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
bec0ee0851 |
Add some boilerplate documentation for IAXVAR and IAXPEER.
........ Merged revisions 355904 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355905 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
2c1b3144cb |
Set the length of the ast_sockaddr, so that we can set it's port later.
Without this, the call to ast_sockaddr_set_port a few lines later is a noop. ........ Merged revisions 355901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355902 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Alec L Davis
|
a4f6d96b2e |
push 'outgoing' flag from sig_XXX up to chan_dahdi
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355851 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
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 |
13 years ago |
Tilghman Lesher
|
a93fbe2ad5 |
Non-verbose output should always go to the remote console, regardless of the previous level.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355749 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
7c373d8c13 |
Pass the correct value to ast_timer_set_rate() for IAX2 trunking.
IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but this value is in milliseconds while ast_timer_set_rate() expects the rate argument to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead. With a default of 20ms, this change makes IAX2 send trunk packets every 20ms instead of every 50ms. Tracked down by myself and Bob Wienholt. ........ Merged revisions 355746 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355747 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355748 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Mark Michelson
|
8a20faa8d7 |
Fix regressions with regards to route-set creation on early dialogs.
This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 ........ Merged revisions 355732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355733 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355745 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Paul Belanger
|
73b5346e79 |
Fix channel opaquification for app_rpt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355667 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
b80fcd77e5 |
Revert a change to audio_audiohook_write_list that had no affect.
When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. ........ Merged revisions 355622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355623 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Richard Mudgett
|
7093cf278c |
Fix compile problem when old version of libvorbisfile v1.1.2 is used.
The principle difference between libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the configure script to detect if libvorbisfile.h declares OV_CALLBACKS_NOCLOSE. * Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile. (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........ Merged revisions 355608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355620 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
7879cccafd |
Fix AMI Monitor action without File header converting channel name into filename.
* Fix potential Solaris crash if Monitor application has a urlbase and no fname_base option. ........ Merged revisions 355574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355575 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
b69fb773d2 |
When IAX2 debugging is enabled, make sure to log 'apathetic' messages too.
........ Merged revisions 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355530 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
45f361c9bd |
Remove IAX_OLD_FIND from chan_iax2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Sean Bright
|
0d12368261 |
Use TRUNK_CALL_START as originally intended.
Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value was also hard-coded in one part of the IAX2 code instead of using the #define. TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but the hard-coded usage was never updated to match. This patch fixes that. ........ Merged revisions 355448 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355449 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Tilghman Lesher
|
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 |
13 years ago |
|
Richard Mudgett
|
51b32041d5 |
Fix voicemail problems when using ogg/vorbis.
Ogg/vorbis was fairly useless as a voicemail file format because it did
not implement the seek and tell format callbacks among other problems.
Since we were already using the libvorbis and libvorbisenc libraries we
can use libvorbisfile as it is also part of the vorbis library package.
* Made use the libvorbisfile to handle the ogg/vorbis file stream. The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
(closes issue ASTERISK-16926)
Reported by: sque
Patches:
ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque
........
Merged revisions 355365 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 355375 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355376 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Richard Mudgett
|
0af15b985c |
Fix lock typo that should be unlock in cel_sqlite_custom reload.
(closes issue ASTERISK-19356)
Reported by: Alex Villacis Lasso
Patches:
asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch (license #5617) patch uploaded by Alex Villacis Lasso
Review: https://reviewboard.asterisk.org/r/1740/
........
Merged revisions 355319 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 355320 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Mark Michelson
|
03894236d0 |
Properly invert the return of a strncmp call.
This was causing identification that should have been made private to be public. (closes issue AST-814) reported by Patrick Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson (license 5430) ........ Merged revisions 355268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355271 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355274 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Jason Parker
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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 |
13 years ago |
|
Sean Bright
|
98111f8f1f |
Clear the high order bit from the destination call number before sending.
send_apathetic_reply takes the incoming frame's source call number as the destination call number for the outgoing frame. If the incoming frame was a full frame, then the high order bit of the source call number is set and will be interpreted as a retransmit when sent back out as the destination call number. ........ Merged revisions 355182 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355183 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355184 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Alexandr Anikin
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002b8bf320 |
call manager_event only if there is not null channel structure
(Closes issue ASTERISK-19298)
Reported by: robinfood
Patches:
issue19298.patch uploaded by may213 (License #5415)
........
Merged revisions 355136 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 355137 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
Russell Bryant
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33322e38f0 |
res_agi: Add AGIEXITONHANGUP variable.
This patch adds a variable AGIEXITONHANGUP for res_agi. If this variable is set to "yes" on a channel, AGI() will exit immediately once a channel hangup has been detected. This was the behavior of AGI() in Asterisk 1.4 and earlier and is still desired by some people. Review: https://reviewboard.asterisk.org/r/1734/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
b978e785dd |
Fix occasional incorrectly delayed call-file execution.
Since the dir timestamp is available at one second resolution, we cannot know if it was updated within the same second after we scanned it. Therefore, we will force another scan if the dir was just modified. * Changed to force another scan if the directory was just modified. (closes issue ASTERISK-19081) Reported by: Knut Bakke Review: https://reviewboard.asterisk.org/r/1688/ ........ Merged revisions 355056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355057 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
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d8af1a4882 |
Fix compile error from most recent ast_channel opaquification installment.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355055 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Joshua Colp
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be2ac1b3dd |
Only allow one 'dialplan reload' to execute at a time as otherwise they would share the same common local context list.
(closes issue AST-758) ........ Merged revisions 355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355010 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355011 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Terry Wilson
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34c55e8e7c |
Opaquify char * and char[] in ast_channel
Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
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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 |
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 |
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Jason Parker
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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
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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
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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 |