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(closes issue #17755)
Reported by: oej
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280549 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17715)
Reported by: vrban
Patches:
17715.patch uploaded by pabelanger (license 224)
Tested by: vrban
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279566 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278957 f38db490-d61c-443f-a65b-d21fe96a405b
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FXS lines normally connect to a telephone. However, when FXS lines are routed
to an external PBX or Key System to act as "external" or "CO" lines, it is
extremely difficult, if not impossible for the external PBX to know when
the call has been disconnected without receiving a polarity reversal on the line.
Now using answeronpolarityswitch and hanguponpolarityswitch keywords that
previously were used only for FXO ports, now applies like functionality for
an FXS port, but from the connected equipment's point of view.
(closes issue #17318)
Reported by: armeniki
Patches:
fxs_linepolarity.diff5.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/797/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278809 f38db490-d61c-443f-a65b-d21fe96a405b
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The documentation for this option did not match the code. Fix that along with
some minor cleanups to the code along the way. Document a slight change in
behavior (to something that was previously undocumented) in UPGRADE.txt.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278425 f38db490-d61c-443f-a65b-d21fe96a405b
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to be used, even when realtime is used.
(closes issue #17082)
Reported by: coolmig
Patches:
20100720__issue17082.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278307 f38db490-d61c-443f-a65b-d21fe96a405b
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There are two changes here:
1. Since the externip setting can now have a port attached
to it, calling it "externip" is misleading. The option is now
documented and parsed as "externaddr." This also extends to the
"matchexterniplocally" setting. It is now documented and parsed
as "matchexternaddrlocally." The old names for the options may
still be used, but they are no longer used in the sip.conf.sample
file.
2. If no port is set for the externaddr, and UDP is the transport
to be used, then we will set the port of the externaddr to that of
the udpbindaddr. This was how things worked prior to the IPv6 merge,
so this is a regression fix.
(closes issue #17665)
Reported by: mmichelson
Patches:
17665.diff#2 uploaded by pprindeville (license 347)
Tested by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277873 f38db490-d61c-443f-a65b-d21fe96a405b
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ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
https://reviewboard.asterisk.org/r/791
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277814 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277028 f38db490-d61c-443f-a65b-d21fe96a405b
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sip.conf configuration for the channel and for devices.
The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.
Review: https://reviewboard.asterisk.org/r/778/
Thanks to dvossel for the review and good advice.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276951 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) | 1 line
Update documentation for voicemail.conf externpass option.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276268 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17628)
Reported by: lmadsen
Tested by: russell, lmadsen
Review: https://reviewboard.asterisk.org/r/774/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275863 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275551 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17600)
Reported by: minaguib
Patches:
ast_unistim_height_v2.patch uploaded by minaguib (license 1078)
Tested by: minaguib
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275466 f38db490-d61c-443f-a65b-d21fe96a405b
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The bridge handling code did not properly consider feature groups when setting
parameters that would affect whether or not a native bridge would be attempted.
If DYNAMIC_FEATURES only include a feature group, a native bridge would occur
that may prevent features from working.
Fix a bug in verbose output that would show the key mapping as empty if it was
using the default mapping and not a custom mapping in the feature group.
Add feature groups to the output of "features show".
Adjust the feature execution logic to match that of the logic when executing
a feature that was not configured through a feature group.
Update features.conf.sample to show that an '=' is still required if using
the default key mapping from [applicationmap].
Finally, clean up a little bit of formatting to better coform to coding
guidelines while in the area.
(closes issue #17589)
Reported by: lmadsen
Patches:
issue_17589.rev4.txt uploaded by russell (license 2)
Tested by: russell, lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275424 f38db490-d61c-443f-a65b-d21fe96a405b
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sample config.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275147 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/678/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274866 f38db490-d61c-443f-a65b-d21fe96a405b
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This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274492 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) | 8 lines
Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
(closes issue #16102)
Reported by: Delvar
Patches:
say.conf.fix.patch uploaded by Delvar (license 908)
(plus a few additional fixes and simplifications by me)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274418 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines
Correct sip.conf.sample comments for prematuremedia option.
(closes issue #17513)
Reported by: festr
Patches:
patch uploaded by festr (license 443)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274316 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17548)
Reported by: cjacobsen
Patches:
say.conf.sample.diff uploaded by cjacobsen (license 1029)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272243 f38db490-d61c-443f-a65b-d21fe96a405b
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This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.
(closes issue #14861)
Reported by: fnordian
Patches:
eventfilter3.patch uploaded by fnordian (license 110),
modified by me
Review: https://reviewboard.asterisk.org/r/673/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271868 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines
Allow users to specify a port for dundi peers.
(closes issue #17056)
Reported by: klaus3000
Patches:
dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271762 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun 2010) | 4 lines
Fixed typo in macro-page
Reported to #asterisk-dev by a student of jsmith.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270987 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15757)
Reported by: Marquis
Patches:
distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
Tested by: Marquis, lmadsen, marcelloceschia
Review: https://reviewboard.asterisk.org/r/351/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270519 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010) | 1 line
Move information about zonemessages into the [zonemessages] section.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270443 f38db490-d61c-443f-a65b-d21fe96a405b
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Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn.
Review: https://reviewboard.asterisk.org/r/696/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270219 f38db490-d61c-443f-a65b-d21fe96a405b
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* chan_dahdi supports dialing configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
False by default. If set, chan_dahdi will ignore failed 'channel' entries.
Handy for the above name-based syntax as it does not depend on
initialization order.
* have my_pri_make_cc_dialstring() only manupulate dial-strings of group
(gGrR) dialing, which make it lsightly more complicated.
https://reviewboard.asterisk.org/r/535/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269238 f38db490-d61c-443f-a65b-d21fe96a405b
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People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.
Patch by snuffy.
(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy
Review: https://reviewboard.asterisk.org/r/461/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269153 f38db490-d61c-443f-a65b-d21fe96a405b
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Update note in sip.conf.sample about externip and externhost with STUN.
(closes issue #16323)
Reported by: klaus3000
Patches:
sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268988 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r268320 | tilghman | 2010-06-04 21:49:52 -0500 (Fri, 04 Jun 2010) | 3 lines
Rest In Peace
http://www.outandaboutnewspaper.com/article/4061
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268321 f38db490-d61c-443f-a65b-d21fe96a405b
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Add the ability to report waiting messages to ISDN endpoints (phones).
Relevant specification: EN 300 650 and EN 300 745
Review: https://reviewboard.asterisk.org/r/599/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267399 f38db490-d61c-443f-a65b-d21fe96a405b
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Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Review: https://reviewboard.asterisk.org/r/568/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267261 f38db490-d61c-443f-a65b-d21fe96a405b
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Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
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This feature generates AMI events in the new aoc event class from the
events passed up by libpri.
Review: https://reviewboard.asterisk.org/r/537/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267008 f38db490-d61c-443f-a65b-d21fe96a405b
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Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.
Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.
Review: https://reviewboard.asterisk.org/r/520/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266926 f38db490-d61c-443f-a65b-d21fe96a405b
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pager messages).
(closes issue #14333)
Reported by: klaus3000
Patches:
20090515__issue14333.diff.txt uploaded by tilghman (license 14)
app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266828 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17392)
Reported by: dps
Patches:
20100525__issue17392.diff.txt uploaded by tilghman (license 14)
Tested by: dps
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265894 f38db490-d61c-443f-a65b-d21fe96a405b
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This commit adds support for calendaring with Exchange Server 2007+ via
Exchange Web Services. Full write support and for querying attendees. Many
thanks to Jan Kaláb for the feature.
(closes issue #17022)
Reported by: pitel
Patches:
res_calendar_ews.c uploaded by pitel (license 1008)
Tested by: pitel, twilson
Review: https://reviewboard.asterisk.org/r/557/
Review: https://reviewboard.asterisk.org/r/668/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265317 f38db490-d61c-443f-a65b-d21fe96a405b
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directmediapermit/directmediadeny support to restrict which peers can do
directmedia based on ip address. In some networks not all phones are fully
routed, i.e. not all phones can ping each other. This patch adds a way to
restrict directmedia for certain peers between certain networks.
(closes issue #16645)
Reported by: raarts
Patches:
directmediapermit.patch uploaded by raarts (license 937)
Tested by: raarts
Review: https://reviewboard.asterisk.org/r/467/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264626 f38db490-d61c-443f-a65b-d21fe96a405b
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Now that Asterisk modules can dynamically create and destroy logger levels
on demand, it's useful to be able to configure a logger channel (console,
file, whatever) to be able to accept log messages from *all* levels, even
levels created dynamically. This patch adds support for this, by allowing
the '*' level name to be used in logger.conf.
Review: https://reviewboard.asterisk.org/r/663/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264160 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17359)
Reported by: alecdavis
Patches:
bug17359.diff.txt uploaded by alecdavis (license 585)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264031 f38db490-d61c-443f-a65b-d21fe96a405b
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From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
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commented out.
This fixes some breakage in the test suite, that uses the contents of asterisk.conf
to discover the install layout on the system.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261364 f38db490-d61c-443f-a65b-d21fe96a405b
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This simply moves the functionality from the Makefile (cleaning it up) into an external
asterisk.conf.samples file. Also updates formatting (easier to read) and grammar
changes to asterisk.conf.samples.
(closes issue #17027)
Reported by: pabelanger
Patches:
0017027.asterisk.conf.v6.patch uploaded by pabelanger (license 224)
Tested by: qwell, lmadsen, pabelanger, chappell
Review: https://reviewboard.asterisk.org/r/616/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261124 f38db490-d61c-443f-a65b-d21fe96a405b
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in all queues.
See the CHANGES file and queues.conf.sample for more details.
(closes issue #17008)
Reported by: jlpedrosa
Patches:
queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002)
Review: https://reviewboard.asterisk.org/r/581/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261051 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17263)
Reported by: pprindeville
Patches:
freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260280 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260148 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines
hidecalleridname parameter in chan_dahdi.conf
Issue #7321 implements a new chan_dahdi configuration option. However, a
change mentioned in the issue was never implemented. This is the change
that will allow the feature to work.
I added a note to chan_dahdi.conf.sample about the feature.
(closes issue #17143)
Reported by: djensen99
Patches:
diff.txt uploaded by djensen99 (license NA) (One line change)
Tested by: djensen99
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259307 f38db490-d61c-443f-a65b-d21fe96a405b
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