https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r300714 | rmudgett | 2011-01-05 14:54:21 -0600 (Wed, 05 Jan 2011) | 21 lines
Merged revision 300711 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) | 14 lines
A call retrieved from hold may wind up with no audio.
If the retrieved call is natively bridged then the call may not have any
audio path. The following warning message is given:
"Failed to add <dfd> to conference <chan>/<chan>: Invalid argument".
* Open the media on a B channel when pri_fixup_principle() moves the call
from a no_b_channel channel to a real channel.
* Added lock protection while pri_fixup_principle() moves a call from one
private structure to another.
* Made some pri_fixup_principle() messages more meaningful.
..........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300716 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r300301 | twilson | 2011-01-04 11:54:41 -0600 (Tue, 04 Jan 2011) | 29 lines
Merged revisions 300298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r300298 | twilson | 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) | 22 lines
Merged revisions 300216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines
Don't authenticate SUBSCRIBE re-transmissions
This only skips authentication on retransmissions that are already
authenticated. A similar method is already used for INVITES. This
is the kind of thing we end up having to do when we don't have a
transaction layer...
(closes issue #18075)
Reported by: mdu113
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: twilson, mdu113
Review: https://reviewboard.asterisk.org/r/1005/
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300302 f38db490-d61c-443f-a65b-d21fe96a405b
Added the moh_signaling option to specify what to do when the channel's
bridged peer puts the ISDN channel on and off of hold.
Implemented as a FSM to control libpri ISDN signaling when the bridged
peer places the channel on and off of hold with the AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD control frames.
JIRA SWP-2687
JIRA ABE-2691
Review: https://reviewboard.asterisk.org/r/1063/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300212 f38db490-d61c-443f-a65b-d21fe96a405b
playing_silence was not initialized with the struct
was initialized, it was being set after the fact
which caused problems if something that relied on
playing_silence being set was called too quickly
(closes issue #18430)
Reported by: stevebrandli
Patches:
externalivr.patch uploaded by thedavidfactor (license 903)
Tested by: thedavidfactor, stevebrandli
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300121 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r300082 | lmadsen | 2011-01-03 07:14:25 -0600 (Mon, 03 Jan 2011) | 11 lines
Increase side of mapping response field.
I've increased the size of the response field in a DUNDi mapping because of
some documentation I'm writing. Previously it was set to AST_MAX_EXTENSION which
is only 80 characters, which is far too small when you're using some dialplan
functions to craft a response. The example I'm using is:
extensions =>
RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300083 f38db490-d61c-443f-a65b-d21fe96a405b
Change order of sending Terminal Capability Set and MasterSlave
Determination packets, MSD send when TCS exchange procedure is done
(we send tcs ack to remote and we have remote tcs ack already
or we receive tcs ack from remote and we have send our tcs ack to
remote already). Some endpoints can work in this sequence only,
i suggest they can't work with both (tcs and msd) exchange procedures
simultaneously.
Also changed StartH245 facility message sending. It send on
incoming calls only due to some endpoints can't proccess properly
this facility messages on their incoming calls.
(closes issue #18433)
Reported by: MrHanMan
Patches:
tcs-msd-h245-3.patch uploaded by may213 (license 454)
Tested by: MrHanMan, may213
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299715 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r299449 | tilghman | 2010-12-22 14:05:02 -0600 (Wed, 22 Dec 2010) | 15 lines
Merged revisions 299448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010) | 8 lines
Resolve warnings by disambiguating the "s" extension as used by chan_dahdi from the "s" extension as used by the AEL macros.
(closes issue #18480)
Reported by: nivek
Patches:
20101215__issue18480__2.diff.txt uploaded by tilghman (license 14)
Tested by: nivek
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299450 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) | 17 lines
Chan_dahdi sends an empty COLP on the bridged channel.
Chan_dahdi always inserts a connected party IE when you call from one
dahdi channel to another dahdi channel, even if no such information was
received on the 2nd channel. This clears the display of many phones.
* Removed leftover artifact from before the valid flag was added.
* Updated all of the channel's caller id information with the new
connected line information instead of just the string parts.
(closes issue #18508)
Reported by: wimpy
Patches:
issue18508_trunk.patch uploaded by rmudgett (license 664)
Tested by: wimpy, rmudgett
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299406 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r299353 | mnicholson | 2010-12-21 09:25:03 -0600 (Tue, 21 Dec 2010) | 30 lines
Merged revisions 299242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r299242 | mnicholson | 2010-12-20 15:25:35 -0600 (Mon, 20 Dec 2010) | 23 lines
Merged revisions 299194,299198,299220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines
Respond as soon as possible with a 202 Accepted to refer requests.
This change also plugs a few memory leaks that can occur when parking sip calls.
ABE-2656
........
r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines
Remove changes to via processing that were not supposed to go into the last commit.
........
r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines
Use ast_free() instead of free()
ABE-2656
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299355 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec 2010) | 20 lines
Fix a couple of CCSS issues.
* Make sure to allocate a cc_params structure
when creating autopeers.
* Use sip_uri_cmp when retrieving SIP CC agents
and monitors in case parameters appear in the
URI.
(closes issue #18504)
Reported by: kkm
(closes issue #18338)
Reported by: GeorgeKonopacki
Patches:
18338.diff uploaded by mmichelson (license 60)
Tested by: GeorgeKonopacki
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299249 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r299131 | tilghman | 2010-12-20 11:47:10 -0600 (Mon, 20 Dec 2010) | 18 lines
Merged revisions 299130 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010) | 11 lines
If a call was not answered, then the billsec was calculated unusually large.
Also, due to a copy and paste error, a request for the answer field would have
given the start value, instead.
(closes issue #18460)
Reported by: joscas
Patches:
20101215__issue18460.diff.txt uploaded by tilghman (license 14)
Tested by: joscas
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299132 f38db490-d61c-443f-a65b-d21fe96a405b
Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation. However, if you used it, it required using different
functions for modifying scheduler contents. This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there. This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.
In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.
Review: https://reviewboard.asterisk.org/r/1007/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299091 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r298960 | tilghman | 2010-12-17 17:52:04 -0600 (Fri, 17 Dec 2010) | 20 lines
Merged revisions 298957 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r298957 | tilghman | 2010-12-17 17:30:55 -0600 (Fri, 17 Dec 2010) | 13 lines
Merged revisions 298905 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines
Let Asterisk find better backtrace information with libbfd.
The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search
for better symbol information within both the Asterisk binary, as well as
loaded modules, to assist when using inline backtraces to track down problems.
Review: https://reviewboard.asterisk.org/r/1055/
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298961 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r298598 | jpeeler | 2010-12-16 14:51:44 -0600 (Thu, 16 Dec 2010) | 21 lines
Merged revisions 298597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r298597 | jpeeler | 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) | 14 lines
Merged revisions 298596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines
Fix improper hangup when doing an attended transfer to queue.
Had to indicate ringing in wait_for_answer so the attended transfer code would
not try and hang up the local channel it created, which would kill the call.
ABE-2624
........
................
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298599 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r298539 | tilghman | 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010) | 8 lines
Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
(closes issue #18464)
Reported by: IgorG
Patches:
realtime_ipv6store.diff uploaded by IgorG (license 20)
(plus a few additional lines by tilghman)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298545 f38db490-d61c-443f-a65b-d21fe96a405b
rmudgett