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r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines
The handling of call transfer signaling for mISDN PTMP is not fully implemented.
The handling of call transfer signaling for mISDN PTMP is not fully
implemented. The signaling of number updates with ISDN/DSS1 ECT
supplementary services (ETS 300 369-1) comes along with a notification
indicator IE and redirection number IE for PTMP. The implementation in
the current Asterisk mISDN channel unfortunately can handle these
information elements only in a NOTIFY message. These information elements
are also signaled in a FACILTY message with a RequestSubaddress facility,
when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of
ETS 300 369-1).
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abe_2526_ast.patch
* Added support to handle the notification indicator IE and redirection
number IE with the RequestSubaddress facility.
* Made misdn_update_connected_line() send a NOTIFY message if Asterisk
originated the call and it is not connected yet.
* Made misdn_update_connected_line() send a FACILITY message if the call
is already connected.
This patch requires the presence of the associated mISDN patches to
compile. I had to enhance mISDN to allow the notification indicator IE
and the redirection number IE to be used with a FACILITY message. Earlier
versions of the Digium enhanced mISDN are no longer going to work.
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abe_2526_misdn.patch
* Made an incoming FACILITY message allow the presence of the notification
indicator IE and the redirection number IE.
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abe_2526_misdnuser_v3.patch
* Added support to send and receive a FACILITY message with the
notification indicator IE and the redirection number IE.
* Added the ability to send a NOTIFY message in PTMP/NT mode to all
responding subcalls in Q.931 states 6, 7, 8, 9, and 25.
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Patches:
abe_2526_ast.patch uploaded by rmudgett (license 664)
abe_2526_misdn.patch uploaded by rmudgett (license 664)
abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
Tested by: rmudgett and reporter
JIRA SWP-2146
JIRA ABE-2526
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r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines
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r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines
Ensure mailbox is not filled to capacity before doing message forwarding.
Specifically, before prompting to record a prepended message the capacity is
checked first. If the mailbox is full the extension will be reprompted.
ABE-2517
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This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
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When originating a call from Unit Under Test to Reference Unit using E&M
RBS signaling mode, I get the following warning message: "Ring/Off-hook in
strange state 3 on channel 1".
Fixed the sig_analog outgoing flag. It was never set when sig_analog was
extracted from chan_dahdi.
JIRA SWP-2191
JIRA AST-408
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IP address of the host we are calling.
This fixes a regression introduced in r274783.
(closes issue #17960)
Reported by: adriavidal
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mich, mnicholson, adriavidal
(closes issue #17676)
Reported by: outcast
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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is used.
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r286757 | mnicholson | 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) | 20 lines
Merged revisions 286756 via svnmerge from
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r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines
Don't clear the username from a realtime database when a registration expires.
Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either.
(closes issue #17551)
Reported by: ricardolandim
Patches:
reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96)
Tested by: ricardolandim, mnicholson
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r286681 | mnicholson | 2010-09-14 13:02:24 -0500 (Tue, 14 Sep 2010) | 14 lines
Merged revisions 286679 via svnmerge from
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r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines
Only drop duplicate answer frames if the channel is bridged.
Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework.
ABE-2473
(related to issue #2342)
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in only one event in calendar
The solution is to use "global" counter of events, since we do new requests for every event and calendar sync after every request. So now we do sync only after last request.
(closes issue #17877)
Review: https://reviewboard.asterisk.org/r/916/
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r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010) | 2 lines
Add documentation on missing backend tables for Voicemail
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r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010) | 2 lines
C precedence got me
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r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010) | 2 lines
Refactor conversion to ast_poll() to fix callparking regression.
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r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | 5 lines
Remove "Internal IP" from sip show settings, as it's not at all useful to display.
(closes issue #17840)
Reported by: oej
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r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör, 11 Sep 2010) | 11 lines
Merged revisions 286267 via svnmerge from
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r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines
Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/
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r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
Merged revisions 286059 via svnmerge from
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r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
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r286117 | pabelanger | 2010-09-10 16:55:06 -0400 (Fri, 10 Sep 2010) | 11 lines
Merged revisions 286114 via svnmerge from
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r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines
Load iax.conf before registering any functions/applications/actions.
Review: https://reviewboard.asterisk.org/r/914/
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r286116 | rmudgett | 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) | 18 lines
Merged revisions 286113 via svnmerge from
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r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines
An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected.
If the ISDN link a pre-connect incoming call is using fails or is reset,
the outgoing leg may not hang up or be delayed in hanging up. (Causes:
PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.)
Just hang up the call if the incoming call leg hangs up before connecting
for any reason. It makes no sense to send a BUSY or CONGESTION control
frame to the outgoing call leg under these circumstances.
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Astdb was determined to be one of the most significant bottlenecks in SIP
registration processing. This patch improved the speed of an astdb load
test by 50000% (yes, Fifty-Thousand Percent). On this particular load test
setup, this doubled the number of SIP registrations the server could handle.
Review: https://reviewboard.asterisk.org/r/825/
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r286024 | tilghman | 2010-09-10 13:30:21 -0500 (Fri, 10 Sep 2010) | 9 lines
Merged revisions 286023 via svnmerge from
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r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines
Missing newline
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r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines
Another fix for Mac OS X.
While trying to fix this the "right" way, I wandered into dependency hell. Two
hours later, I backed out, and just removed the offending code. ast_inline_api
only goes one level deep and then it breaks. Ouch.
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r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
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r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
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r285818 | pabelanger | 2010-09-09 18:49:19 -0400 (Thu, 09 Sep 2010) | 15 lines
Merged revisions 285817 via svnmerge from
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r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines
GCC 4.2.x optimizations result in improper behavior of GSM codec
(closes issue #17688)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347)
Tested by: mkeuter, pprindeville
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r285744 | qwell | 2010-09-09 15:09:23 -0500 (Thu, 09 Sep 2010) | 16 lines
Merged revisions 285742 via svnmerge from
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r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines
Transmit silence when reading DTMF in ast_readstring.
Otherwise, you could get issues with DTMF timeouts causing hangups.
(closes issue #17370)
Reported by: makoto
Patches:
channel-readstring-silence-generator.patch uploaded by makoto (license 38)
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r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
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r280812 | tilghman | 2010-08-03 15:52:20 -0500 (Tue, 03 Aug 2010) | 16 lines
Merged revisions 280811 via svnmerge from
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r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) | 9 lines
Prevent DAHDI channels from overriding the callerid, once it's been set by the user.
(closes issue #16661)
Reported by: jstapleton
Patches:
20100414__issue16661.diff.txt uploaded by tilghman (license 14)
20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: jstapleton
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specified item.
Matches up with FIELDQTY() and CUT().
(closes issue #17713)
Reported by: gareth
Patches:
svn-279754.diff uploaded by gareth (license 208)
Tested by: gareth, tilghman
Review: https://reviewboard.asterisk.org/r/810/
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(closes issue #17663)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
diff2 uploaded by sperreault (license 252)
get_domain.diff uploaded by sperreault (license 252)
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(closes issue #17737)
Reported by: oej
Patches:
doc.diff uploaded by sperreault (license 252)
Tested by: mmichelson
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Review: https://reviewboard.asterisk.org/r/836/
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r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010) | 2 lines
Document -B and -W flags and regenerate manpage from sgml
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r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 Aug 2010) | 2 lines
Allow the pipe, but also allow the comma
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a 1 somewhere.
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Compensate for this.
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r280556 | tilghman | 2010-07-29 16:07:03 -0500 (Thu, 29 Jul 2010) | 7 lines
Off-by-one error
(closes issue #17590)
Reported by: atis
Patches:
20100729__issue17590.diff.txt uploaded by tilghman (license 14)
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populate the FAXOPT output variables.
FAX-222
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r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) | 11 lines
fixes wrong SRV query for TLS connection
(closes issue #17612)
Reported by: marcelloceschia
Patches:
chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079)
chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079)
Tested by: marcelloceschia, st, pabelanger
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(closes issue #17755)
Reported by: oej
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(closes issue #17751)
Reported by: b11d
Patches:
strdupa_oops.diff uploaded by malcolmd (license 924)
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r280449 | dvossel | 2010-07-29 14:05:25 -0500 (Thu, 29 Jul 2010) | 18 lines
Merged revisions 280448 via svnmerge from
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r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines
fixes issue with translator frame not getting freed
A translator frame even if it local storage so the translation path
can be freed. This issue prevented g729 licenses from being freed up.
(closes issue #17630)
Reported by: manvirr
Patches:
encoder_fix.diff uploaded by dvossel (license 671)
Tested by: manvirr, dvossel
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r280345 | jeang | 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) | 10 lines
Merged revisions 280341 via svnmerge from
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r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines
Fix a dsp structure leak occuring when a local channel is put into a meetme
conference, then masquaraded away.
ABE-2422
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related to r280302
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r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines
Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported.
ABE-2229
Review: https://reviewboard.asterisk.org/r/813/
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Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699.
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