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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Update changes for SIPRemoveHeader()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@168575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@168265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@167792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13836)
Reported by: eliel
Patches:
chan_dahdi.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@165433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@165180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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to poke all peers at seemingly the same time.
(closes issue #13217)
Reported by: cervajs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13698)
Reported by: fhackenberger
Patches:
20081211__bug13698.diff.txt uploaded by Corydon76 (license 14)
Tested by: fhackenberger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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to AMI Redirect.
Review: http://reviewboard.digium.com/r/89/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Based on cli_permissions.conf configuration file, we are able to permit or deny
cli commands based on some patterns and the local user and group running rasterisk.
(Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/
(closes issue #11123)
Reported by: eliel
Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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until the originating line hangs up.
(closes issue #12382)
Reported by: one47
Patches:
zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23)
zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463)
Tested by: fleed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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res_timing_timerfd
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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documentation for more information on how to use it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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remotesecret => our password for a remote service
secret => our authentication when someone calls us
Secret => still has both functions if remotesecret is not used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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had no effect
* Updated dialing API documentation to indicate that timeouts
are specified in milliseconds
* Added a new timeout argument to the Page application. If time
expires, any endpoints which have not answered will be hung up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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UPDATE query initially
affects 0 rows.
(closes issue #13083)
Reported by: Corydon76
Patches:
20081031__bug13083.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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the low default value of featuredigittimeout, I decided it
was high time to change it. I have changed the default to
2000 ms based on a suggestion from Leif Madsen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13734)
reported and patched by: jvandal
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13731)
reported and patched by: caio1982
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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devices and lines.
(closes issue #13412)
Reported by: wedhorn
Patches:
config-restruct-v4.diff uploaded by wedhorn (license 30)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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for more details of this command.
(closes issue #13326)
Reported by: ib2
Patches:
bug13326_trunk_20080822.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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when Asterisk was configured using --prefix and 'make samples' is run
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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to mean that no timeout is desired.
(closes issue #13625)
Reported by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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without needing to respecify an extension pattern multiple times.
(closes issue #13632)
Reported by: blitzrage
Patches:
20081006__bug13632.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage, Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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making the new/old message count available to local devices. (issue #AST-77)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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and "leavewhenempty" options are configured in queues.conf.
Instead of using vague terms like "yes," "no," "loose," and
"strict," we now accept a comma-separated list of values
to determine when to consider a member available.
Extended details can be found in the queues.conf.sample
file. Note also that the above four referenced values are
still accepted for backwards-compatibility, but are mapped
internally to the new method of representing the option.
AST-105
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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call-id in the dialog-info event package used with extension state subscriptions
on Snom phones. Then, when the phone sends an INVITE with Replaces for the
special callid, Asterisk will perform a pickup on the extension that was
subscribed to.
The original code on this issue was submitted by xylome. However, contributions
have been made by (at least) mgernoth and pkempgen. The final patch was written
by seanbright, and includes the necessary logic to allow this work in a
technology independent way.
(closes issue #5014)
Reported by: xylome
Patches:
issue5014-trunk.diff uploaded by seanbright (license 71)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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