This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so.
(closes issue #14179)
Reported by: CrashHD
Tested by: CrashHD
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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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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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
is only found if it dialed the extension that was subscribed to. You can now
specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as
it's value implies, ignore the current context of the caller when doing the
lookup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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
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lying about what's possible. The tz cannot be set in a
context like this. It can only be set in the general
section or per-mailbox.
Thanks to sasargen on #asterisk-dev for pointing this out
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines
The documentation listed the ability to set 'maxmsg' per
context. The truth is that you can only set this in the general section
or per mailbox. Thus I am updating the sample config file to be more
accurate.
Thanks to sasargen on IRC for bringing up this issue.
........
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the calling party when subscribed to the state of an extension that is ringing.
This has some limitations which are documented in sip.conf.sample.
(closes issue #13827)
Reported by: seanbright
Patches:
issue13827.patch uploaded by seanbright (license 71)
Reviewed by: russellb
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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
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines
A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying
to get the one-touch parking stuff working, because it
didn't occur to me that I had to also have the corresponding
options in the dial command! Duh! (In all this time, I never
set this up before!)
So, to keep some poor fool from suffering the same fate,
I made the features.conf.sample file mention the corresponding
opts in dial/queue; and the docs for dial/app specifically
mention the corresponding decls in the feature.conf file.
I hope this doesn't spoil some vast, eternal plan...
........
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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
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