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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188378 f38db490-d61c-443f-a65b-d21fe96a405b
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This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
(closes issue #3450)
Reported by: cmaj
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188342 f38db490-d61c-443f-a65b-d21fe96a405b
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The F option to app_dial has been modified to accept no parameters and perform
the above functionality. I don't see anywhere else that is doing function
overloading, but this really is the best place for this operation because:
- It makes it close to the 'g' option in the argument list which provides
similar functionality.
- The existing code to support the current F option provides a very
convienient location to add this new feature.
(closes issue #12381)
Reported by: michael-fig
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187491 f38db490-d61c-443f-a65b-d21fe96a405b
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a dialplan variable.
This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.
(closes issue #13243)
Reported by: samdell3
Patches:
13243.diff uploaded by file (license 11)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186624 f38db490-d61c-443f-a65b-d21fe96a405b
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Asterisk.
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14695)
Reported by: mneuhauser
Patches:
cause-for-hangup-manager-action.patch uploaded by mneuhauser (license 425)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185704 f38db490-d61c-443f-a65b-d21fe96a405b
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Added an option to respond to a SIP invite with only the single most preferred joint codec. This limits the options of what codecs the other side can use.
(closes issue #12485)
Reported by: bamby
Review: http://reviewboard.digium.com/r/206/
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Review: http://reviewboard.digium.com/r/199/
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Progress DTMF was added into app_dial's D() option. In CHANGES it should have been updated under 1.6.3 rather than 1.6.2.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182607 f38db490-d61c-443f-a65b-d21fe96a405b
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The D() option in app_dial is only able to send DTMF after the call has been answered. A progress option has been added to D() to allow DTMF to be sent upon receiving PROGRESS. This allows DTMF to be sent before the call is answered.
(closes issue #12123)
Reported by: VoipForces
Patches:
app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419)
dtmf_progress.patch uploaded by dvossel (license 671)
Tested by: VoipForces, dvossel
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182362 f38db490-d61c-443f-a65b-d21fe96a405b
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This commit introduces official support for R2 signaling in chan_dahdi. The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1)
are using it in each of the following countries: Colombia, Nepal, Thailand,
Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.
The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.
I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message. These are the names that I
could find in the mantis issue.
(closes issue #12509)
Reported by: moy
Patches:
chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen
Review: http://reviewboard.digium.com/r/40/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
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(actually its $(localstatedir)/run/asterisk
Makes setups with asterisk as non-root easier to manage because you can
setup permissions on this dir instead of touching a file and setting
permissions on that.
Files that come to mind are asterisk.pid and asterisk.ctl socket.
Prodded by and ok @russell
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This commit brings in the bridging core, bridging technologies,
and the ConfBridge application.
For usage information on the ConfBridge application please see
the output of "core show application ConfBridge" from the CLI.
For API documentation please see the doxygen page describing the
architecture and the documentation for each API call.
Review: http://reviewboard.digium.com/r/93/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180369 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13826)
Reported by: azielke
Patches:
pickupsound2-trunk.patch uploaded by azielke (license 548)
__20081124_bug_13826_updated.patch uploaded by lmadsen (license 10)
Tested by: lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178919 f38db490-d61c-443f-a65b-d21fe96a405b
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what kind of encryption is enabled as well.
Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.
(closes issue #14427)
Reported by: snuffy
Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
Review: http://reviewboard.digium.com/r/173/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178300 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14372)
Reported by: fhackenberger
Patches:
voicemail_individual_subject_and_body_1.6.1 uploaded by fhackenberger (license 592)
with additional fixes by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178107 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177320 f38db490-d61c-443f-a65b-d21fe96a405b
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time ago.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177291 f38db490-d61c-443f-a65b-d21fe96a405b
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From a user point-of-view, this adds new CLI commands and Manager Actions to
better facilitate the reloading of queues and the resetting of their statistics.
The new CLI commands are the "queue reload" and "queue reset stats" commands.
The new manager actions are the QueueReload and QueueReset commands.
Review: http://reviewboard.digium.com/r/115
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This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well.
(closes issue #13285)
Reported by: sgofferj
Tested by: russell
Review: http://reviewboard.digium.com/r/150/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175344 f38db490-d61c-443f-a65b-d21fe96a405b
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This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well.
(closes issue #14266)
Reported by: jcovert
Patches:
chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
iax.conf.sample.patch uploaded by jcovert (license 551)
Tested by: jcovert, dvossel
Review: http://reviewboard.digium.com/r/143/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@174046 f38db490-d61c-443f-a65b-d21fe96a405b
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potential issues
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173047 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13877)
Reported by: eliel
Patches:
cli_manager_logout.patch.txt uploaded by eliel (license 64)
Tested by: eliel, putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173028 f38db490-d61c-443f-a65b-d21fe96a405b
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reviewboard this before committing again...
reopened 11583 until all Russell's issues are
resolved.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172929 f38db490-d61c-443f-a65b-d21fe96a405b
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to be used within the dial app, before a call is bridged.
Many thanks to sobomax for submitting this patch.
Quoting from bug 11582:
"So the goal of the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function is not well suited
for this purpose as it uses much call bridge specific data and doesn't separate a
detection of feature from a feature handler call. So a new function ast_feature_detect()
has been extracted off the ast_feature_interpret() function but keeping the original
logic intact except some insignificant changes to locking.
"Having created the ast_feature_detect() function the possibility to use feature detection
in almost any place of the asterisk code. So a call to this function has been added to
wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler
however and uses old call leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature currently is the only one
supported during call setup as other features as call parking and call transfer don't make much
sense during call setup. However if need in some of the features would arise it is much easier to
implement as the infrastructure changes are already in place with this patch."
I have cleaned up the patch somewhat, and verified that the existing functionality is not
harmed, and that the new functionality works. Terry has committed his stuff, and there were
no conflicts (see 14274).
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license 359)
patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax (license 359)
11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172890 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
........
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(closes issue #14301)
Reported by: amorsen
review: http://reviewboard.digium.com/r/128/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@171757 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168760 f38db490-d61c-443f-a65b-d21fe96a405b
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Update changes for SIPRemoveHeader()
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168639 f38db490-d61c-443f-a65b-d21fe96a405b
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sip.conf
With this commit, a register => line in sip.conf may contain a port number in the
"user" section of the line. Please see CHANGES and sip.conf.sample for more
details regarding this.
(closes issue #14198)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168575 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13065)
Reported by: tzafrir
Patches:
sip_nat_settings uploaded by tzafrir (license 46)
sip_nat_settings_6 uploaded by mvanbaak (license 7)
Tested by: tzafrir, pabelanger, Dovid and moi
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168265 f38db490-d61c-443f-a65b-d21fe96a405b
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This patch adds the functionality to app_queue of calculating
the average amount of time that channels are bridged for a
queue. The algorithm used to calculate the average is the same
exponential average currently used to calculate the average holdtime.
See the CHANGES file to see the methods you may use to view this
information.
(closes issue #13960)
Reported by: coolmig
Patches:
app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167792 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13836)
Reported by: eliel
Patches:
chan_dahdi.c.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167791 f38db490-d61c-443f-a65b-d21fe96a405b
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This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor
continue recording the call even after the transfer
has completed.
It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.
(closes issue #13538)
Reported by: mbit
Patches:
13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/102/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166092 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14075)
Reported by: rcasas
Patches:
app_originate.c uploaded by rcasas (license 641), heavily modified by me
Tested by: russell
Review: http://reviewboard.digium.com/r/95/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165433 f38db490-d61c-443f-a65b-d21fe96a405b
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enabled (either globally or for a specific peer), chan_sip will treat any SDP
data it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session version
received is different from the current SDP session version. This option is
required to interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with Microsoft OCS
which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/
(closes issue #13958)
Reported by: toc
Tested by: toc
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165180 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14028)
Reported by: mostyn
Patches:
timezone-v2.patch uploaded by mostyn (license 398)
(with additional code guideline fixes and a memory leak fix by me - license 14)
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to poke all peers at seemingly the same time.
(closes issue #13217)
Reported by: cervajs
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164809 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13698)
Reported by: fhackenberger
Patches:
20081211__bug13698.diff.txt uploaded by Corydon76 (license 14)
Tested by: fhackenberger
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to AMI Redirect.
Review: http://reviewboard.digium.com/r/89/
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SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND.
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after T38 is negotiated.
Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/
This functionality is for issue AST-140.
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