the settings show up and reflect
accurately. Changed settings, I see the new settings after restart.
- rnewton
On April 8, 2015, 5:52 p.m., Kevin Harwell wrote:
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configuration option to any
appropriate place, such as the iax.conf.sample file.
- rnewton
On April 3, 2015, 9:32 p.m., Y Ateya wrote:
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.
Calls Internal to voicemail and other features.
External to internal DID calls.
External to internal feature calls.
Basically tried to call as many ways as I could through all the various
features. Everything seemed to work.
Thanks,
rnewton
https://reviewboard.asterisk.org/r/4504/#comment25422
Make sure to add a timing interface, we could use res_timing_timerfd.so
- rnewton
On March 16, 2015, 5:48 p.m., Jonathan Rose wrote:
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to add a timing interface to modules.conf. res_timing_timerfd.so
is probably fine.
- rnewton
On March 16, 2015, 5:48 p.m., Jonathan Rose wrote:
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this in the diff I just E-mailed over to you.
- rnewton
On March 16, 2015, 3:37 p.m., Jonathan Rose wrote:
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On March 24, 2015, 10:39 p.m., rnewton wrote:
/trunk/configs/basic-pbx/extensions.conf, lines 16-30
https://reviewboard.asterisk.org/r/4503/diff/1/?file=72533#file72533line16
These should move to the External-Features context and get their
extensions changed to the full DID
, but you'll see
them in the review.
- rnewton
On March 24, 2015, 9:53 p.m., rnewton wrote:
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On March 24, 2015, 10:41 p.m., rnewton wrote:
Tested with 4488, only a few modifications made. Emailed you a diff with
the changes and external extensions.
When testing this patch with 4488 I ran through the following tests:
*patch1 - internal stuff*
internal user to internal user (audio
On March 24, 2015, 10:32 p.m., rnewton wrote:
Tested with 4488. Pretty much worked fine.
You will need to add a timing interface to modules.conf.
res_timing_timerfd.so is probably fine.
When testing with 4488 I ran through the following tests:
*patch1 - internal stuff*
internal user
a diff with the
changes and external extensions.
- rnewton
On March 16, 2015, 3:37 p.m., Jonathan Rose wrote:
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patch I've run into
some one-way audio and issues where in certain calling scenarios MOH does not
play. I'm investigating that further tomorrow.
- rnewton
On March 16, 2015, 3:37 p.m., Jonathan Rose wrote:
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with these with outside connectivity this morning, so we'll
see if anything needs to change.
- rnewton
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connectivity this
morning, so we'll see if anything needs to change.
- rnewton
On March 16, 2015, 5:48 p.m., Jonathan Rose wrote:
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extensions - and place the
actual internal extensions into some other context.
rnewton wrote:
I'm changing Pre-Internal to Internal and Internal to General as
the contexts included there are not really specific to internal use.
Hah I forgot about the default general context. That caused me
On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
/branches/13/configs/basic-pbx/extensions.conf, lines 135-136
https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line135
I'm assuming we're going to replace the prompts eventually? :-)
rnewton wrote:
Yeah I
.
- rnewton
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On March 13, 2015, 2:32 p.m., rnewton wrote
extensions - and place the
actual internal extensions into some other context.
I'm changing Pre-Internal to Internal and Internal to General as the
contexts included there are not really specific to internal use.
- rnewton
.
- rnewton
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On March 13, 2015, 2:32 p.m., rnewton wrote
features. Everything seemed to work.
Thanks,
rnewton
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://reviewboard.asterisk.org/r/4488/#comment25236
Blob, oops.
/branches/13/configs/basic-pbx/logger.conf
https://reviewboard.asterisk.org/r/4488/#comment25237
This snuck in there during troubleshooting apparently. I should comment
that back out.
- rnewton
On March 13, 2015, 2:32 p.m
and retrieved them
with all users. Verified MWI working with all phones.
Thanks,
rnewton
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the pattern match here and for hints as for the
current users we only need '_11XX' .
- rnewton
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: https://reviewboard.asterisk.org/r/4379/diff/
Testing
---
Setup Asterisk with configuration, connected up three phones using the first
three users. Made calls between them all, left voicemails and retrieved them
with all users. Verified MWI working with all phones.
Thanks,
rnewton
.
- rnewton
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On Feb. 13, 2015, 12:46 a.m., rnewton wrote
://reviewboard.asterisk.org/r/4379/#comment25024
Forgot to remove this Verbose call that I was using for debugging an issue
when testing.
- rnewton
On Feb. 13, 2015, 12:46 a.m., rnewton wrote:
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prefer lining up on the priority, but I'm usually writing very short
dialplans for testing issues.
I played with both and I agree with you here. I think lining up on the '=' will
be better as this dialplan grows.
- rnewton
fundamental stuff which isn't applicable
to the wide world.
rnewton wrote:
I used MAC addresses as that is what we use as an example in our security
best practices document:
http://svnview.digium.com/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt?view=markup
Perhaps
all, left voicemails and retrieved them
with all users. Verified MWI working with all phones.
Thanks,
rnewton
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://reviewboard.asterisk.org/r/4379/#comment24995
Re-phrase this and remove trailing whitespace.
- rnewton
On Feb. 13, 2015, 12:46 a.m., rnewton wrote:
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'=' in dialplan.
I'm not sure why.
Joshua Colp wrote:
I'm the same way. Across the dialplans I've seen they have primarily used
'='. If this uses '=' I'd actually expect people to not know it's the same
and question it.
rnewton wrote:
I've seen both, '=' may be more common
keep it as
an actual file. Also, this example set is not intended to be laid over the
other samples, so there will be no file to symlink to.
- rnewton
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On Jan. 30, 2015, 7:51 p.m., Paul Belanger wrote:
/branches/13/configs/examples/super_awesome_company/indications.conf, line 1
https://reviewboard.asterisk.org/r/4379/diff/1/?file=71110#file71110line1
Again, this file seem to be the default. I'd rather see a symlink.
rnewton
addresses.
With either of those approaches we should be able to use the less secure
extension numbered auth users.
What would be the issues either of these approaches other than an attacker on
the internal network?
- rnewton
mention on the second. I
just remembered the 's' option for voicemailmain as well to have it ignore the
pin.
- rnewton
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to have these sounds installed as well.
It was already on there. :)
- rnewton
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you to reuse code between applications. However, this
example does not do this, it could be possible to create generate
subroutines in asterisk samples to share across my examples.
I ended up rewriting this whole section for a few reasons.
- rnewton
it.
Oops. I left these in here. I originally added them as I wasn't able to get
things working without loading them (warnings/errors complained). I'll
investigate what the issue was.
- rnewton
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them
with all users. Verified MWI working with all phones.
Thanks,
rnewton
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connections. A new one
will always be created. This is an issue being tracked at URL.
I think you forgot a URL there.
- rnewton
On Jan. 12, 2015, 1:33 p.m., Joshua Colp wrote:
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to the title.
You could also link some keywords to other content on the wiki. That is always
helpful in-case someone lands on this page but they are not aware of some other
helpful content related to the topic.
Otherwise, ship it!
- rnewton
On Jan. 12, 2015, 1:33 p.m., Joshua Colp wrote
On Jan. 12, 2015, 5:32 p.m., rnewton wrote:
It looks detailed enough for me.
One thing I might change would be some of the headings and the page title.
You have a couple sub-headings reading PJSIP Transport Selection,
whereas the others do not mention PJSIP. When you find one
@ 5:52PM CDT - Builds with no issues in dev-mode.
Thanks,
rnewton
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419821
Diff: https://reviewboard.asterisk.org/r/3854/diff/
Testing
---
Once finalized I'll build in dev-mode with it to make sure I didn't screw up
any tags.
7/30/14 @ 5:52PM CDT - Builds with no issues in dev-mode.
Thanks,
rnewton
clarity.
Diffs (updated)
-
/branches/1.8/main/manager.c 419821
Diff: https://reviewboard.asterisk.org/r/3854/diff/
Testing
---
Once finalized I'll build in dev-mode with it to make sure I didn't screw up
any tags.
Thanks,
rnewton
/r/3854/#comment23315
I think I need literal tags for the mentions of underscores here and just
below.
- rnewton
On July 30, 2014, 10:40 p.m., rnewton wrote:
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.
This is a slight modification to improve clarity.
Diffs (updated)
-
/branches/1.8/main/manager.c 419821
Diff: https://reviewboard.asterisk.org/r/3854/diff/
Testing
---
Once finalized I'll build in dev-mode with it to make sure I didn't screw up
any tags.
Thanks,
rnewton
- Builds with no issues in dev-mode.
Thanks,
rnewton
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with
'_', the variable will be inherited into channels created from the current
channel. If the variable name is prefixed with '__', the variable will be
inherited into channels created from the current channel and all children
channels.
- rnewton
://reviewboard.asterisk.org/r/3854/diff/
Testing
---
Once finalized I'll build in dev-mode with it to make sure I didn't screw up
any tags.
Thanks,
rnewton
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Testing
---
Once finalized I'll build in dev-mode with it to make sure I didn't screw up
any tags.
Thanks,
rnewton
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)
-
/branches/1.8/main/manager.c 419562
Diff: https://reviewboard.asterisk.org/r/3854/diff/
Testing
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Once finalized I'll build in dev-mode with it to make sure I didn't screw up
any tags.
Thanks,
rnewton
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,
rnewton
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in
; the general context.
- rnewton
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416556
Diff: https://reviewboard.asterisk.org/r/3621/diff/
Testing
---
Sample file update, doesn't affect configuration. Only rearranged text, no
addition or removal of options or contexts. So, no testing, other than looking
at it!
Thanks,
rnewton
416556
Diff: https://reviewboard.asterisk.org/r/3622/diff/
Testing
---
Sample file update, doesn't affect configuration. Only rearranged text, no
addition or removal of options or contexts. So, no testing, other than looking
at it!
Thanks,
rnewton
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Ship it!
Works for me.
- rnewton
On June 18, 2014, 1:29
parking
I used the groups, Pickup Options, Transfer Options and Parking Options. I
didn't feel there was enough options to justify sub-groups beyond that. Thanks!
- rnewton
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, no testing, other than looking
at it!
Thanks,
rnewton
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Sample file update, doesn't affect configuration. Only rearranged text, no
addition or removal of options or contexts. So, no testing, other than looking
at it!
Thanks,
rnewton
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or contexts. So, no testing, other than looking
at it!
Thanks,
rnewton
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: https://reviewboard.asterisk.org/r/3621/diff/
Testing
---
Sample file update, doesn't affect configuration. Only rearranged text, no
addition or removal of options or contexts. So, no testing, other than looking
at it!
Thanks,
rnewton
update, doesn't affect configuration. Only rearranged text, no
addition or removal of options or contexts. So, no testing, other than looking
at it!
Thanks,
rnewton
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for trunk
https://reviewboard.asterisk.org/media/uploaded/files/2014/06/11/0eb22a12-e297-4ec3-8191-fd1774a50f33__coreshowhints_trunk.patch
Thanks,
rnewton
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Thanks,
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- rnewton
On June 11, 2014, 6 p.m., Scott Griepentrog wrote:
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for naming Asterisk 12+ CLI aliases?
Those syntax backwards compatibility templates are commented out in the config
sample file. Only the friendly template is enabled. No system should be
dependent on those.
- rnewton
://reviewboard.asterisk.org/r/3572/diff/
Testing
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Tested the added aliases.
Thanks,
rnewton
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that
reloading chan_pjsip doesn't parse config. Remembering pjsip reload is just
easier.
Diffs
-
/branches/12/configs/cli_aliases.conf.sample 414779
Diff: https://reviewboard.asterisk.org/r/3572/diff/
Testing
---
Tested the added aliases.
Thanks,
rnewton
/Updating+or+Upgrading+Asterisk
- rnewton
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On May 15, 2014, 9:13 p.m
On May 16, 2014, 5:24 p.m., Scott Griepentrog wrote:
Nitpick: There is an extra space in the Asterisk support life-cycle.
Fixed thanks
- rnewton
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edit access feel free to edit typos/logic errors
straight-away, otherwise just report them on here.
Diffs
-
Diff: https://reviewboard.asterisk.org/r/3542/diff/
Testing
---
Thanks,
rnewton
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Diffs
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Diff: https://reviewboard.asterisk.org/r/3542/diff/
Testing
---
Thanks,
rnewton
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://reviewboard.asterisk.org/r/3542/diff/
Testing
---
Thanks,
rnewton
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Attachments
Patch for 11,12,Trunk
https://reviewboard.asterisk.org/media/uploaded/files/2014/04/17/4421ac84-bf9f-417d-9449-a582ee90430d__asterisk23550_11plus.patch
Thanks,
rnewton
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understandable and less ambiguous than the original?
Diffs
-
/branches/12/formats/format_wav.c 407338
Diff: https://reviewboard.asterisk.org/r/3188/diff/
Testing
---
Thanks,
rnewton
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this documentation patch clarifies things.
Diffs
-
/branches/12/configs/pjsip.conf.sample 407338
Diff: https://reviewboard.asterisk.org/r/3180/diff/
Testing
---
Only adding informational text to the pjsip.conf.sample file.
Thanks,
rnewton
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Ship it!
Ship It!
- rnewton
On Nov. 2, 2013, 1:30 p.m
On Feb. 5, 2014, 7:37 p.m., rnewton wrote:
Ship It!
Tested command, output looks good and makes sense to me.
- rnewton
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Ship it!
Ship It!
- rnewton
On Feb. 3, 2014, 8:18 p.m
sure this documentation patch clarifies things.
Diffs (updated)
-
/branches/12/configs/pjsip.conf.sample 407338
Diff: https://reviewboard.asterisk.org/r/3180/diff/
Testing
---
Only adding informational text to the pjsip.conf.sample file.
Thanks,
rnewton
to format_wav.c ?
2) Is the message more understandable and less ambiguous than the original?
Diffs
-
Diff: https://reviewboard.asterisk.org/r/3188/diff/
Testing
---
Thanks,
rnewton
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understandable and less ambiguous than the original?
Diffs (updated)
-
/branches/12/formats/format_wav.c 407338
Diff: https://reviewboard.asterisk.org/r/3188/diff/
Testing
---
Thanks,
rnewton
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Testing
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Only adding informational text to the pjsip.conf.sample file.
Thanks,
rnewton
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me make sure this documentation patch clarifies things.
Diffs (updated)
-
/branches/12/configs/pjsip.conf.sample 407338
Diff: https://reviewboard.asterisk.org/r/3180/diff/
Testing
---
Only adding informational text to the pjsip.conf.sample file.
Thanks,
rnewton
this line
manually, but I'll poke the script to see if we can get the comment alignment
better later on.
- rnewton
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to break anything.
Thanks,
rnewton
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anything.
Thanks,
rnewton
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Location State Application(Data)
PJSIP/6002-0001 (None) Up AppDial((Outgoing Line))
PJSIP/6001- (None) Up Dial(PJSIP/6002,15)
2 active channels
1 active call
1 call processed
- rnewton
On Dec. 20, 2013
the Channel ID to it's left.
- rnewton
On Dec. 20, 2013, 4:24 a.m., George Joseph wrote:
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On Dec. 20, 2013, 7:35 p.m., rnewton wrote:
For pjsip show endpoints, a formatting issue:
Endpoint/CID
State. Channels.
I/OAuth:
AuthId/UserName...
Aor
On Dec. 20, 2013, 7:35 p.m., rnewton wrote:
For pjsip show endpoints, a formatting issue:
Endpoint/CID
State. Channels.
I/OAuth:
AuthId/UserName...
Aor
: 6002/sip:6002@10.24.18.138:5060;ob Unknown
nan
In this example, both 6001 and 6002 are registered, but 6001 has made a call
and then hung up. 6002 has not made a call yet.
- rnewton
On Dec. 20, 2013, 4:24 a.m., George Joseph wrote
On Dec. 20, 2013, 7:35 p.m., rnewton wrote:
For pjsip show endpoints, a formatting issue:
Endpoint/CID
State. Channels.
I/OAuth:
AuthId/UserName...
Aor
.
Diffs
-
branches/12/configs/pjsip.conf.sample 404396
Diff: https://reviewboard.asterisk.org/r/3086/diff/
Testing
---
Only documentation outside of source code, doesn't appear to break anything.
Thanks,
rnewton
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