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:
>
> ---
> This is an automatically generated e-mail. To reply, visit
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:
>
> ---
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asteris
work.
Thanks,
rnewton
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email 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,
rn
t catch. That is an artifact from testing the config. I forgot to
uncomment it back. Thanks!
- rnewton
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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 t
> 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
diff with the
changes and external extensions.
- rnewton
On March 16, 2015, 3:37 p.m., Jonathan Rose wrote:
>
> ---
> This is an automatically generated e-mail. To reply, visit:
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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 the
You'll note this in the diff I just E-mailed over to you.
- rnewton
On March 16, 2015, 3:37 p.m., Jonathan Rose wrote:
>
> ---
> This is an automatically generated e-mail. To reply, visit:
> https:
onfigs/basic-pbx/modules.conf
<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:
>
> --
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:
>
> ---
> This is an automatically generated e-mail. To reply, visit
, but you'll see
them in the review.
- rnewton
On March 24, 2015, 9:53 p.m., rnewton wrote:
>
> ---
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.as
calls.
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,
rn
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:
>
> ---
> This
ff gets ran when an internal context
> calls into an extension that is just included by Internal.
>
> Matt Jordan wrote:
> Hm. That's a fair point.
>
> I'd rename this slightly then: have the Pre-Internal be "Internal" -
> since it should be a
ow.
I'm finally testing with these with outside connectivity this morning, so we'll
see if anything needs to change.
- rnewton
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https://revie
tside connectivity this
morning, so we'll see if anything needs to change.
- rnewton
On March 16, 2015, 5:48 p.m., Jonathan Rose wrote:
>
> ---
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboa
> 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
could be used within the operation of a Gosub and that any
Return would Return out of the current Gosub. I probably misunderstood
something fundamental about Gosubs.
- rnewton
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ht
> Don't you need a config file for this?
Forgot to 'svn add' cdr.conf and cdr_custom.conf. Whoops.
- rnewton
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applied to all "internal" 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
tps://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, 20
rious
features. Everything seemed to work.
Thanks,
rnewton
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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
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me/asterisk.conf PRE-CREATION
/branches/13/configs/examples/awesome/README PRE-CREATION
Diff: 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 an
ttern match?
Nope. Fixed. I also adjusted the pattern match here and for hints as for the
current users we only need '_11XX' .
- rnewton
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tps://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:
>
> ---
> This is a
console => verbose,notice,warning,error
That works.
- rnewton
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-----
> > That being said, I'm also fine if we line things up based on priority -
> > but I'd go for a consistent scheme and stick with it. (I do think lining
> > things up on priority leads to a lot of leading white space, but that's
> > just my laziness k
tps://reviewboard.asterisk.org/r/4379/#comment24995>
Re-phrase this and remove trailing whitespace.
- rnewton
On Feb. 13, 2015, 12:46 a.m., rnewton wrote:
>
> ---
> This is an automatically generated e-mail. To reply,
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
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nes, and Digium phones allow separate user and
> authuser to be specified.
>
> Joshua Colp wrote:
> Your statement is true but it would be nice if we could err on the side
> of not falling into a trap of doing fundamental stuff which isn't applicable
> to the
'=>', but I (for some reason) still prefer
> > '=>' 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
efault. I'd rather see a symlink.
>
> rnewton wrote:
> For the sake of avoiding confusion I'll 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.
I forgot I did change this as w
ile=7#file7line1>
> >
> > Seems to be the default file, rather see a symlink
It isn't the default file. For the sake of avoiding confusion I'll 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
/extensions.conf, lines
> > 53-66
> > <https://reviewboard.asterisk.org/r/4379/diff/1/?file=71109#file71109line53>
> >
> > I disagree with this comments. Moving subroutines into there own
> > contexts allow you to reuse code
722 as well. On installation it's
> > wise to have these sounds installed as well.
It was already on there. :)
- rnewton
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https://
change the spec to specify that IT has locked down traffic between
Asterisk and the public internet to only allow inbound traffic from the ITSP
addresses.
Or, on Asterisk we can use ACL's to limit traffic allowed to the internal
network and ITSP addresses.
With either o
otely by employees who dial 256-555-1234. When
>employees dial voicemail remotely, they must input both their mailbox number
>and their pin code.
Aha. I read the first line, but missed the "pin code" mention on the second. I
just remembered the 's' option for voicemai
in a separate GoSub.
Yeah I ended up rewriting that whole section for several reasons.
> On Jan. 27, 2015, 8:34 p.m., Matt Jordan wrote:
> > /branches/13/configs/examples/super_awesome_company/modules.conf, lines
> > 103-108
> > <https://revie
hree 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
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> 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",
> &
as adding "PJSIP" 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
sting connections. A new one
will always be created. This is an issue being tracked at ."
I think you forgot a URL there.
- rnewton
On Jan. 12, 2015, 1:33 p.m., Joshua Colp wrote:
>
> ---
> This is an automatically generated e-ma
tags.
7/30/14 @ 5:52PM CDT - Builds with no issues in dev-mode.
Thanks,
rnewton
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ger.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.
7/30/14 @ 5:52PM CDT - Builds with no issues in dev-mode.
T
14 @ 5:52PM CDT - Builds with no issues in dev-mode.
Thanks,
rnewton
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nction
calls.
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 scr
tps://reviewboard.asterisk.org/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:
>
> ---
> This is an automatically generated e-mail.
t 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 scr
hanges:
"This command can be used to set the value of channel variables or dialplan
functions. When setting variables, if the variable name is prefixed with
'_', the variable will be inherited into channels created from the current
channel. If the variable name
dated)
-
/branches/1.8/main/manager.c 419562
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
--
___
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
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/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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hanks,
rnewton
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hanks,
rnewton
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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,
rn
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,
rn
ed a note under the "Parking Options" heading.
; These options apply to all parking lots, including the default lot defined in
; the general context.
- rnewton
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---
Ship it!
Works for me.
- rnewton
On June 18, 2014, 1:29
s. So, no testing, other than looking
at it!
Thanks,
rnewton
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hanks,
rnewton
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with a
> > blank line separating the groups:
> > pickup
> > transfer
> > transfer-sounds
> > atxfer
> > parking
I used the groups, Pickup Options, Transfer Options and Parking Options. I
didn't feel there was enough options to justify
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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: 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,
rn
ns or contexts. So, 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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File Attachments
patch 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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Presence:available Watchers 1
1 hint matching extension 6002
- rnewton
On June 11, 2014, 6 p.m., Scott Griepentrog wrote:
>
> ---
> This is an autom
91-fd1774a50f33__coreshowhints_trunk.patch
Thanks,
rnewton
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; Do you have an alternative to propose 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
-
es_pjsip.so" and if some are like me; I constantly forget 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
anches/12/configs/cli_aliases.conf.sample 414779
Diff: https://reviewboard.asterisk.org/r/3572/diff/
Testing
---
Tested the added aliases.
Thanks,
rnewton
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have wiki 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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> 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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.org/wiki/display/AST/Updating+or+Upgrading+Asterisk
- rnewton
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---
iff: https://reviewboard.asterisk.org/r/3542/diff/
Testing
---
Thanks,
rnewton
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Diffs
-
Diff: https://reviewboard.asterisk.org/r/3542/diff/
Testing
---
Thanks,
rnewton
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B, IT.
File 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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Thanks,
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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,
rn
he message content accurate according to format_wav.c ?
2) Is the message more 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
--
:
1) Is the message content accurate according to format_wav.c ?
2) Is the message more understandable and less ambiguous than the original?
Diffs (updated)
-
/branches/12/formats/format_wav.c 407338
Diff:
extension.\n"
Please let me know:
1) Is the message content accurate according 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
--
_
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,
rn
---
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---
Ship it!
Ship It!
- rnewton
On Feb. 3, 2014, 8:18 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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---
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---
Ship it!
Ship It!
- rnewton
On Nov. 2, 2013, 1:30 p.m
help 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,
rn
07338
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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> On Dec. 20, 2013, 7:46 p.m., rnewton wrote:
> > The state reflected in the command outputs remains "Invalid" and does not
> > update until the endpoint has made at least one call. Then it will change
> > to "Not in use"
> >
>
> On Dec. 20, 2013, 7:35 p.m., rnewton wrote:
> > For "pjsip show endpoints", a formatting issue:
> >
> >
> >
> > I/OAuth:
> >
> > Aor:
> >
> > Contact:
> >
> >Id
2
Contact: 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.
> On Dec. 20, 2013, 7:35 p.m., rnewton wrote:
> > For "pjsip show endpoints", a formatting issue:
> >
> >
> >
> > I/OAuth:
> >
> > Aor:
> >
> > Contact:
> >
> >Id
> On Dec. 20, 2013, 7:35 p.m., rnewton wrote:
> > For "pjsip show endpoints", a formatting issue:
> >
> >
> >
> > I/OAuth:
> >
> > Aor:
> >
> > Contact:
> >
> >Id
xten: 6002 CLCID: "" <>
For the "Channel:" line, the "State" and "Time(sec)" values are out of their
columns and the State is pushed up against the Channel ID to it's left.
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, 4:24 a.m., George Joseph wrote:
>
> ---
break anything.
Thanks,
rnewton
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7;t appear to break anything.
Thanks,
rnewton
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