https://origsvn.digium.com/svn/asterisk/trunk
........
r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 lines
Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
(closes issue #14322)
Reported by: amessina
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r172400 | rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 lines
channels/chan_dahdi.c
* Added doxygen comments to the major dahdi structures.
* Fixed PRI and SS7 using an incorrect string value if the extension
delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as long
as the stripmsd option.
* Fixed PRI not handling unknown TON/NPI prefix letters correctly.
* Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample
* Updated some documentation.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r166382 | mmichelson | 2008-12-22 15:08:03 -0600 (Mon, 22 Dec 2008) | 44 lines
Merged revisions 166380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines
Fix a deadlock relating to channel locks and autoservice
It has been discovered that if a channel is locked prior
to a call to ast_autoservice_stop, then it is likely that
a deadlock will occur. The reason is that the call to
ast_autoservice_stop has a check built into it to be sure
that the thread running autoservice is not currently trying
to manipulate the channel we are about to pull out of
autoservice.
The autoservice thread, however, cannot advance beyond where
it currently is, though, because it is trying to acquire
the lock of the channel for which autoservice is attempting
to be stopped.
The gist of all this is that a channel MUST NOT be locked
when attempting to stop autoservice on the channel.
In this particular case, the channel was locked by a call
to ast_read. A call to ast_exists_extension led to autoservice
being started and stopped due to the existence of dialplan
switches.
It may be that there are future commits which handle the same
symptoms but in a different location, but based on my looks through
the code, it is very rare to see a construct such as this one.
(closes issue #14057)
Reported by: rtrauntvein
Patches:
14057v3.patch uploaded by putnopvut (license 60)
Tested by: rtrauntvein
Review: http://reviewboard.digium.com/r/107/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008) | 13 lines
Merged revisions 152215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) | 6 lines
Inherit ALL elements of CallerID across a local channel.
(closes issue #13368)
Reported by: Peter Schlaile
Patches:
20080826__bug13368.diff.txt uploaded by Corydon76 (license 14)
........
................
r152287 | jpeeler | 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines
Merged revisions 152286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) | 2 lines
Buffer policy setting for half is not needed.
........
................
r152369 | tilghman | 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) | 15 lines
Merged revisions 152368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines
Reset all DIAL variables back to blank, in case Dial is called multiple times
per call (which could otherwise lead to inconsistent status reports).
(closes issue #13216)
Reported by: ruddy
Patches:
20081014__bug13216.diff.txt uploaded by Corydon76 (license 14)
Tested by: ruddy
........
................
r152467 | tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10 lines
Merged revisions 152463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) | 3 lines
Quoting in the wrong direction
(Fixes AST-107)
........
................
r152569 | russell | 2008-10-29 00:34:26 -0500 (Wed, 29 Oct 2008) | 15 lines
Merged revisions 152539 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) | 7 lines
Fix an incorrect usage of sizeof()
(closes issue #13795)
Reported by: andrew53
Patches:
chan_sip_sizeof.patch uploaded by andrew53 (license 519)
........
................
r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) | 22 lines
Merged revisions 152538 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines
A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying
to get the one-touch parking stuff working, because it
didn't occur to me that I had to also have the corresponding
options in the dial command! Duh! (In all this time, I never
set this up before!)
So, to keep some poor fool from suffering the same fate,
I made the features.conf.sample file mention the corresponding
opts in dial/queue; and the docs for dial/app specifically
mention the corresponding decls in the feature.conf file.
I hope this doesn't spoil some vast, eternal plan...
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) | 9 lines
Merged revisions 147517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines
If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8)
........
................
r147689 | kpfleming | 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines
Merged revisions 147681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines
when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected)
........
................
r148000 | tilghman | 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines
Merged revisions 147997 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) | 4 lines
When blank, callerid name and number should display "unknown caller" in voicemail
emails.
(Closes issue #13643)
........
................
r148112 | mmichelson | 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines
Merged revisions 146026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines
(closes issue #13579)
Reported by: dwagner
(closes issue #13584)
Reported by: dwagner
Tested by: murf, putnopvut
The thought occurred to me that the res= from the extension spawn
was ending up being returned from the bridge.
"Thou shalt not poison the return value". Made the change
and it appears to allow blind xfers to work as normal.
If I'm wrong, reopen the bugs. But it looks good to me!
Many thanks to putnopvut for helping me reproduce this!
........
................
r148268 | tilghman | 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines
Merged revisions 148257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) | 7 lines
User not notified of temporary greeting, if ODBC storage is in use.
(closes issue #13659)
Reported by: moliveras
Patches:
20081009__bug13659.diff.txt uploaded by Corydon76 (license 14)
Tested by: moliveras
........
................
r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) | 11 lines
Merged revisions 148916 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines
Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used
in headers like 'Subject' and 'To'.
Closes AST-107.
........
................
r148988 | tilghman | 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines
Merged revisions 148987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines
Some compilers warn, some don't. Fixing.
........
................
r149062 | tilghman | 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) | 13 lines
Merged revisions 149061 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines
Check correct values in the return of ast_waitfor(); also, get rid of a
possible memory leak.
(closes issue #13658)
Reported by: explidous
Patch by: me
........
................
r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) | 15 lines
Merged revisions 149130 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines
Don't allow reserved characters to be used in register
lines in sip.conf.
(closes issue #13570)
Reported by: putnopvut
........
................
r149201 | mmichelson | 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines
Merged revisions 149200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines
Update the queue with the correct number of calls and
whether the call was completed within the service level
when a transfer takes place. This way, we do not "break"
the leastrecent and fewestcalls strategies by not logging
a call until after the transferred call has ended.
(closes issue #13395)
Reported by: Marquis
Patches:
app_queue.c.transfer.patch uploaded by Marquis (license 32)
........
................
r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines
Merged revisions 149204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines
Add a tolerance period for sync-triggered audiohooks
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.
Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet
........
................
r149208 | mmichelson | 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines
Merged revisions 149207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines
Call register_peer_exten even in the case that the peer's
IP/port does not change.
(closes issue #13309)
Reported by: dimas
Patches:
v2-13309.patch uploaded by dimas (license 88)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r138311 | jpeeler | 2008-08-15 18:46:09 -0500 (Fri, 15 Aug 2008) | 20 lines
Merged revisions 138119,138151,138238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) | 4 lines
Fixes the dahdi restart functionality. Dahdi restart allows one to restart all DAHDI channels, even if they are currently in use. This is different from unloading and then loading the module since unloading requires the use count to be zero. Reloading the module is different in that the signalling is not changed from what it was originally configured. Also, this fixes not closing all the file descriptors for D-channels upon module unload (which would prevent loading the module afterwards).
(closes issue #11017)
........
r138151 | jpeeler | 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line
declared static mutexes using AST_MUTEX_DEFINE_STATIC macro
........
r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) | 1 line
initialize condition variable ss_thread_complete using ast_cond_init
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@138333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r137848 | tilghman | 2008-08-14 11:52:43 -0500 (Thu, 14 Aug 2008) | 17 lines
Merged revisions 137847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) | 9 lines
When creating the secondary subchannel name, it is necessary to compare to
the existing channel name without the "Zap/" or "DAHDI/" prefix, since our
test string is also without that prefix.
(closes issue #13027)
Reported by: dferrer
Patches:
chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525)
(Slightly modified by me, to compensate for both names)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@137849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r133041 | mmichelson | 2008-07-23 12:54:03 -0500 (Wed, 23 Jul 2008) | 15 lines
Merged revisions 133038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines
Small cleanup. Move the declaration of the DAHDI_SPANINFO
variable to the block where it is used. This allows one
less #ifdef HAVE_PRI to clutter things up.
Thanks to Tzafrir for pointing this out on #asterisk-dev
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r132721 | kpfleming | 2008-07-22 16:21:56 -0500 (Tue, 22 Jul 2008) | 14 lines
Merged revisions 132712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul 2008) | 6 lines
ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense'
(closes issue #12160)
Reported by: tzafrir
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) | 6 lines
Add configuration option to chan_dahdi.conf to allow buffering policy and number of buffers to be configured per channel. Syntax:
buffers=<num of buffers>,<policy>
Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate".
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul 2008) | 12 lines
Merged revisions 130039 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines
add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today
(related to issue #13042)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@130041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r128122 | mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line
Add option to wait to be able to explicitly send ACM via the Proceeding() application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@128124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | 38 lines
Merged revisions 127663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines
The CDRfix4/5/6 omnibus cdr fixes.
(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror
(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11
(closes issue #11849)
Reported by: greyvoip
As to 11849, I think these changes fix the core problems
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.
Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.
(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@127830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r125333 | kpfleming | 2008-06-26 10:50:07 -0500 (Thu, 26 Jun 2008) | 13 lines
Merged revisions 125327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines
ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number
(issue AST-80)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) | 18 lines
Merged revisions 125132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@125146 65c4cc65-6c06-0410-ace0-fbb531ad65f3