Age | Commit message (Collapse) | Author | Files | Lines |
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Reported by: thetatag
Patch by: thetatag/stevens/tilghman
Closes issue #5331
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89078 f38db490-d61c-443f-a65b-d21fe96a405b
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- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
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(closes issue #11171, reported and patched by blitzrage)
Many thanks!
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Closes issue #9972
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines
Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable.
(closes issue #11006)
Reported by: pj
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 lines
Fix improbable but possible memory leaks in chan_zap.
(closes issue #11166)
Reported by: eliel
Patches:
chan_zap.c.patch uploaded by eliel (license 64)
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way.
(closes issue #11167)
Reported by: eliel
Patches:
chan_agent.c.patch uploaded by eliel (license 64)
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(closes issue #11168)
Reported by: IgorG
Patches:
unistimjb-88863-1.patch uploaded by IgorG (license 20)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88935 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines
After seeing crashes related to channel variables, I went looking around at the
ways that channel variables are handled. In general, they were not handled in
a thread-safe way. The channel _must_ be locked when reading or writing from/to
the channel variable list.
What I have done to improve this situation is to make pbx_builtin_setvar_helper()
and friends lock the channel when doing their thing. Asterisk API calls almost
all lock the channel for you as necessary, but this family of functions did not.
(closes issue #10923, reported by atis)
(closes issue #11159, reported by 850t)
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Remove unset variables, they would be misleading.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines
When traversing the list of channel variables here in transmit_invite(), the
asterisk channel must be locked, as this data may change at any time.
(I have seen numerous reports of crashes related to the handling of channel
variables. There are a couple of issues on the bug tracker related to it,
but it has also been noted on IRC and mailing lists. So, I am finding and
fixing some places where channel variables are handled improperly.)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) | 2 lines
Fix up some indentation.
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r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 lines
If a SIP channel is put on hold multiple times do not keep incrementing the onHold value.
(closes issue #11085)
Reported by: francesco_r
Tested by: blitzrage
(closes issue #10474)
Reported by: acennami
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11163)
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r88585 | qwell | 2007-11-05 11:19:41 -0600 (Mon, 05 Nov 2007) | 4 lines
Make sure we destroy the config structure on configuration failure.
Issue 11163, patch by eliel.
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(closes issue #11158)
Reported by: eliel
Patches:
chan_unistim.c.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88510 f38db490-d61c-443f-a65b-d21fe96a405b
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details and examples are in include/asterisk/stringfields.h.
Not applicable to older branches except for 1.4 which will
receive a fix for the routines that free memory pools.
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This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones. The following models have been confirmed
to work: i2002, i2004 and i2050.
(closes issue #8864)
Reported by: c_hans
Patches:
chan_unistim.patch uploaded by c (license 304)
ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88368 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 lines
Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context).
(closes issue #10702)
Reported by: slavon
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 lines
If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive.
(closes issue #5747)
Reported by: chandi
Patches:
infofix-81430-1.patch uploaded by IgorG (license 20)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4 lines
Make sure we set the poll fds to NULL after free()ing it.
Part of issue 11017, patch by tzafrir.
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(closes issue #11138)
Reported by: eliel
Patches:
pbx_dundi.c.patch uploaded by eliel (license 64)
chan_sip.c.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88077 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: eliel
Patch by: eliel
Closes issue #11135
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11130)
(closes issue #11132)
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r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines
Don't try to allocate memory that we're just going to re-allocate later anyways.
Issues 11130 and 11132, patch by eliel.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) | 11 lines
Merge the changes from team/russell/iax2_poke_fix and iax2-poke-fix-trunk
There was a race condition related to the handling of POKEing peers. Essentially,
a reference to a peer is held by the scheduler when there are pending callbacks,
but the reference count didn't reflect it. So, it was possible for a peer to hit
a reference count of zero and have its destructor begin to be called at the same
time that the scheduler thread ran a POKE related callback. If that happened,
a crash would likely occur.
(closes issue #11082, closes issue #11094)
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r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1 line
Only try to clean out h323/ if the h323/Makefile exists.
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r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6 lines
Fix issue where if both sides of the dialog cancelled the dialog at the same time chan_sip could kepe retransmitting a response for no reason.
(closes issue #9566)
Reported by: atca_pres
Patches:
bug9566.patch uploaded by oej
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releases yet. chan_misdn should now build against the latest release.
(closes issue #11103)
Reported by: IgorG
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(issue #11096)
Patches:
chan_agent.c.patch uploaded by eliel (license 64)
chan_local.c.patch uploaded by eliel (license 64)
chan_features.c.patch uploaded by eliel (license 64)
chan_zap.c.patch uploaded by eliel (license 64)
res_monitor.c.patch uploaded by eliel (license 64)
res_realtime.c.patch uploaded by eliel (license 64)
res_crypto.c.patch uploaded by eliel (license 64)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11079)
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r86982 | qwell | 2007-10-24 15:56:47 -0500 (Wed, 24 Oct 2007) | 5 lines
Correctly respect hidecalleridname configuration option.
Simplify code slightly in the process.
Issue 11079, reported by ddv2005
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didn't make much sense
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r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4 lines
After reading online I have confirmed that Record-Route headers should be copied to 1xx responses as well.
(closes issue #10113)
Reported by: makoto
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and it's called "make". :)
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accomplish any good and causes problems for plenty of people...
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r86598 | crichter | 2007-10-22 11:21:15 +0200 (Mo, 22 Okt 2007) | 1 line
we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan does not match after an overlap call. Also added out_cause=1
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CallForwarding and so forth
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Closes issue #11039, as suggested by seanbright.
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Also fixes a few cli messages and some minor formatting.
(closes issue #11001)
Reported by: seanbright
Patches:
newcli.1.patch uploaded by seanbright (license 71)
newcli.2.patch uploaded by seanbright (license 71)
newcli.4.patch uploaded by seanbright (license 71)
newcli.5.patch uploaded by seanbright (license 71)
newcli.6.patch uploaded by seanbright (license 71)
newcli.7.patch uploaded by seanbright (license 71)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 lines
Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote.
(closes issue #11027)
Reported by: ramonpeek
Patches:
11027-1.diff uploaded by ramonpeek (license 266)
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r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4 lines
Set port number in received as information for registrations as well.
(closes issue #11028)
Reported by: brad-x
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(patch made by seanbright, pointed out in #asterisk-dev on IRC)
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r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | 8 lines
If Asterisk is in the middle of shutting down, respond to OPTIONS
with 503 Unavailable.
(closes issue #10994)
Reported by: eserra
Patches:
sip-options-503.patch uploaded by eserra (license 45)
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