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: http://svn.digium.com/svn/asterisk/trunk@153904 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@153223 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@152807 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@150311 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@146640 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@145226 f38db490-d61c-443f-a65b-d21fe96a405b
This will print the subs and their status for every line (if any).
wedhorn did most of the work with his patch which introduced
'skinny show debug' but a discussion on IRC stated that it should be
added to 'skinny show lines'
Input on the output format by Qwell on IRC.
(closes issue #13344)
Reported by: wedhorn
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140938 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: erousseau
This was a reasonable enhancement request, which was
easy to implement. Since it's an enhancement, it
could only be applied to trunk.
Basically, for accounting where "initiated" seconds
are billed for, if the microseconds field on the end
time is greater than the microseconds field for the
answer time, add one second to the billsec field.
The implementation was requested by erousseau, and
I've implemented it as requested. I've updated the
CHANGES, the cdr.conf.sample, and the .h files
accordingly, to accept and set a flag for the
corresponding new option. cdr.c adds in the extra
second based on the usec fields if the option is
set. Tested, seems to be working fine.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140057 f38db490-d61c-443f-a65b-d21fe96a405b
This set of changes enhances IAX2 encryption support by adding key rotation
to provide enhanced security. The key used for encryption is rotated right
after the call gets set up, and then again every few minutes. This was
discussed at the last AstriDevCon. For interoperability with older versions
of Asterisk, there is an option that disables key rotation.
(closes issue #13018)
Reported by: bbryant
Patches:
07072008__iax2_key_rotation.diff uploaded by bbryant (license 36)
Tested by: russell, bbryant
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135158 f38db490-d61c-443f-a65b-d21fe96a405b
driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
implementations. Asterisk has, for a long time,
had its own implementation of poll(2) which
just used the input arguments to call select(2).
In 1.4, this internal implementation was used
for Darwin systems. This was removed in Asterisk
trunk at some point, but it seems as though this
was not the right move to make.
On Mac OS X, it appears as though the poll used
to gather CLI input does not respond properly
when connecting via a remote Asterisk console.
Reverting to the use of Asterisk's poll fixed
the issue.
Also, there is now an option for the configure
script, --enable-internal-poll, which will allow
for anyone to use Asterisk's internal poll
implementation in case they suspect that their
system's poll implementation is buggy.
closes issue #11928)
Reported by: adriavidal
Patches:
1.6.0-configurev2.patch uploaded by putnopvut (license 60)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134125 f38db490-d61c-443f-a65b-d21fe96a405b
fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130951 f38db490-d61c-443f-a65b-d21fe96a405b
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127558 f38db490-d61c-443f-a65b-d21fe96a405b
to how language is handled between two channels whose native
language is different. Prior to this patch, app_dial would have
the callee inherit the caller's language, and app_queue would not.
After this patch, app_dial no longer has the language inheritance
capability. This seems to make the most sense since it seems more
natural for a person to hear files played back in his/her native
language instead of the language of the person on the far end of
the call. See the CHANGES file for hints on how to keep the
previous behavior of app_dial if desired.
(closes issue #12489)
Reported by: bcnit
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125647 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line
Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122128 f38db490-d61c-443f-a65b-d21fe96a405b
tilghman