https://origsvn.digium.com/svn/asterisk/trunk
................
r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines
Merged revisions 221769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines
Occasionally losing use of B channels in chan_misdn.
I have not been able to reproduce the problem of losing channels.
However, I have seen in the code a reentrancy problem that might give
these symptoms.
The reentrancy patch does several things:
1) Guards B channel and B channel structure allocation.
2) Makes the B channel structure find routines more precise in locating records.
3) Never leave a B channel allocated if we received cause 44.
The last item may cause temporary outgoing call problems, but they should
clear when the line becomes idle.
(closes issue #15490)
Reported by: slutec18
Patches:
issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: rmudgett, slutec18
(closes issue #15458)
Reported by: FabienToune
Patches:
issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: FabienToune, rmudgett, slutec18
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@221871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines
Remove ability to control T.38 FAX error correction from udptl.conf.
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@221602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines
Merged revisions 221360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines
Fix SRV lookup and Request-URI generation in chan_sip.
This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur.
(closes issue #14418)
Reported by: klaus3000
Tested by: klaus3000, mnicholson
Review: https://reviewboard.asterisk.org/r/369/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@221478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@221302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines
Merged revisions 220873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines
Reduce CPU usage related to building a peer merely for devicestates.
This fixes a 100% CPU problem in the SIP driver, found by profiling
the driver while the problem was occurring.
(closes issue #14309)
Reported by: pkempgen
Patches:
20090924__issue14309.diff.txt uploaded by tilghman (license 14)
Tested by: pkempgen, vrban
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@220998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines
Make deletion of temporary greetings work properly with IMAP_STORAGE
When imapgreetings was set to yes, the message was being deleted but wasn't
actually being expunged. When imapgreetings was set to no, the file based
message was not being deleted at all. All good now!
(closes issue #14949)
Reported by: noahisaac
Patches:
vm_tempgreeting_removal.patch uploaded by noahisaac (license 748),
modified by me
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@220835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r220718 | jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
Fix building of registration entry in build_peer when using callbackextension
Check for remotesecret option was unintentionally always true, which therefore
caused the secret option to never be used. Thanks to dvossel for pointing out
the exact fix.
(closes issue #15943)
Reported by: tpsast
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@220720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009) | 3 lines
add name argument for the CALLERID dialplan function to the xml documentation.
Pointed out to me on IRC by snuff-home. Thanks
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@220631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r220028 | mvanbaak | 2009-09-24 10:36:18 +0200 (Thu, 24 Sep 2009) | 14 lines
Merged revisions 220027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) | 7 lines
mkpkgconfig does not need bash so make it use /bin/sh
This fixes building on all systems that don't have bash
at /bin/bash
Reported by _ys on #asterisk-dev
Tested by _ys on #asterisk-dev
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@220030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r219895 | lmadsen | 2009-09-23 12:46:46 -0500 (Wed, 23 Sep 2009) | 13 lines
Add Mantis work flow documention.
This commit adds the doxygen changes that I've made to describe the Mantis
work flow documentation for the open source issue tracker. This should make
it easier to determine the flow of issues through the issue tracker, and what
those statuses mean.
(closes issue #15902)
Reported by: lmadsen
Patches:
mantisworkflow.h uploaded by lmadsen (license 10)
Review: https://reviewboard.asterisk.org/r/367/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines
Merged revisions 219816 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines
When IMAP variables were changed during a reload, Voicemail did not use the new values.
This change introduces a configuration version variable, which ensures that
connections with the old values are not reused but are allowed to expire
normally.
(closes issue #15934)
Reported by: viniciusfontes
Patches:
20090922__issue15934.diff.txt uploaded by tilghman (license 14)
Tested by: viniciusfontes
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r219520 | dvossel | 2009-09-18 18:20:58 -0500 (Fri, 18 Sep 2009) | 15 lines
Merged revisions 219519 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines
iax2 frame double free
The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines
Merged revisions 219450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines
via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's invite_branch
variable must be updated so CANCEL requests use the correct branch id.
(closes issue #15262)
Reported by: maniax
Patches:
asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608)
invite_new_branch_trunk.diff uploaded by dvossel (license 671)
Tested by: maniax, dvossel
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) | 27 lines
Merged revisions 219303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines
INVITE w/Replaces deadlock fix
This patch cleans up the locking logic in chan_sip.c's
handle_invite_replaces() function as well as making use
of ast_do_masquerade() rather than forcing the masquerade
on an ast_read(). The code had several redundant unlocks
that would result in 'freed more times than we've locked!'
errors. I cleaned these up as well as moving all the unlock
logic to the end of the function. This patch should also
resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls.
(closes issue #15151)
Reported by: irroot
Patches:
invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
Tested by: irroot, dvossel
Review: https://reviewboard.asterisk.org/r/371/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines
Merged revisions 219136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316)
Reported by: vmarrone
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines
On TCP and TLS connections do not attempt to stop retransmission of the packet internally.
This was preventing responses from being properly processed because the packet was not being found
causing handle_response to return prematurely.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009) | 20 lines
Merged revisions 218867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines
Fixes CID pattern matching behavior to mirror that of extension pattern matching.
Pattern matching for extensions uses a type of scoring system, giving values for
specificity to each character in the pattern. Unfortunately, this is done character
by character, in order. This does lead to some less specific patterns being first
in line for matching, but it will usually get the job done.
This patch merely brings CID matching to the same level as extension matching.
This patch does not attempt to tackle the problem shared by extension matching.
(closes issue #14708)
Reported by: klaus3000
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009) | 16 lines
Merged revisions 218798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines
Remove the IAXy firmware from Asterisk.
The firmware can now be found on downloads.digium.com, where the rest of our
binary downloads live. This was the last part of our Asterisk tarballs that
was considered non-free by Debian. :-)
(closes issue #15838)
Reported by: paravoid
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
