On Mon, Nov 28, 2016, at 11:39 AM, Matt Riddell wrote:
>
> > On 27/11/2016, at 6:44 PM, Joshua Colp <jc...@digium.com> wrote:
> >
> > On Wed, Nov 23, 2016, at 06:41 PM, Matt Riddell wrote:
> >>
> >> There doesn't appear to be a way to moni
le would not be present. You can
subscribe to all channels by subscribing to "channel:" using the
/applications//subscription resource, and "bridge:" for all
bridges.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL
l have to hold up updating the FreeBSD Asterisk port.
FreeBSD isn't really a platform we support or actively target, so we'll
see what we can do and if it should hold up the release.
[1] https://issues.asterisk.org/jira
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - H
ot support exchanging
SDP like this. A SIP proxy would be a better fit.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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mentation though[2]. Are there any in particular you are interested
in?
[1] https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime
[2] https://issues.asterisk.org/jira/browse/ASTERISK-26572
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsvi
DOCUMENTATION","1478458346.826",""
> 5007-004b","SIP/truck1-004c","Dial","SIP/truck1/6052736,80","2016-11
> -06 18:54:02",,"2016-11-06 18:54:23",20,0,"NO
> ANSWER","DOCUMENTATION","1478458442.
al numbers
>
> exten => _9XXX,n,Set(CALLERID(all)="$CallerID" <3818008000>)
>
> exten => _9XXX,n,Dial(SIP/voip-truck/1381${EXTEN:1},80)
This should also work fine. You'll need to provide the console output of
an attempt. Different logic may be executed.
-
both .conf and in realtime? Do you also
have chan_sip loaded?
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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_
ructures into JSON, and then back.
There's nothing currently available that I know of for JSON in the
dialplan.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at:
updated from 11.21.2 to 11.24.0. System is shutting itself down when
hitting a depreciated dialplan command:
This appears to be a regression introduced in 11.23.0 actually, and is
applicable to those two operations. If you create an issue I can throw a
patch up.
--
Joshua Colp
Digium, Inc
sion
+++ b/ChangeLog
+++ /dev/null
+++ /dev/null
+++ b/asterisk-11.24.0-summary.html
+++ b/asterisk-11.24.0-summary.txt
Please file an issue on the issue tracker[1].
[1] https://issues.asterisk.org/jira
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville
j run wrote:
any git tag in particular? (13.12.0-rc1 is my best guess)
Yes, that is the current release candidate.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
jrun wrote:
any updates as to when this would be available for 13?
The version of 13 with the changes required to support it is currently
in release candidate status. I expect it to reach release status soon.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
ct
to localhost won't work. You will either have to bind to 0.0.0.0 or
connect to 192.168.1.8 instead.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Matt Riddell wrote:
On 19/10/2016, at 2:32 AM, Luca Pradovera <luca.pradov...@gmail.com
<mailto:luca.pradov...@gmail.com>> wrote:
Would UnicastRTP be able to output u-law frames directly? If so, I
think that is all I need.
Joshua Colp did a great writeup that may work for yo
Matt Riddell wrote:
On 17/10/2016, at 4:07 PM, Joshua Colp <jc...@digium.com
<mailto:jc...@digium.com>> wrote:
Matt Riddell wrote:
On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradov...@gmail.com
<mailto:luca.pradov...@gmail.com>
<mailto:luca.pradov...@gmail.c
t I'm saving to a file and then sending that file once
recording has finished.
The UnicastRTP channel driver allows you to send RTP to a specific
target address with media. Combined with Chanspy (or Snoop channels in
ARI) you can duplicate audio from a channel and send it off to where you
want.
--
? It should be user-configurable.
There is no ability to do this. It's generated by PJSIP itself.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
issues? That can cause chan_sip to
lock up for a period of time.
[1]
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check
er.
I've fixed it, thanks!
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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it gets the data).
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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people are at
AstriCon presently and the repos are generally unmaintained. This does
mean that the repos do NOT receive security updates.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-26407
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806
mentioned
and manually manipulate things. This is also, I think, the first time
I've ever heard of a company wanting it to behave precisely like that.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com
Please file an
issue[1].
[1] https://issues.asterisk.org/jira
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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it somewhat and moved back into Asterisk.
I've created an issue[1] to clean this up, both the upgrade and the
sample config.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-26389
-- Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
noticed.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Dmitry Melekhov wrote:
09.09.2016 14:08, Dmitry Melekhov пишет:
09.09.2016 13:45, Joshua Colp пишет:
Dmitry Melekhov wrote:
Hello!
Upgraded 13.10 to 13.11.1 today and now I see messages in log:
[Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request
'REGISTER' from '"
sers though
Could you tell me what is this?
It means that the request could not be matched to an endpoint, just like
the message says :D this could be because endpoints in pjsip.conf could
not be loaded or if from a database, they couldn't be loaded from there.
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Joshua Colp
Digium, Inc.
ntinue to be called. Is something
upstream from Asterisk doing the voicemail?
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.as
does the console show?
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Joshua Colp
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erring to is not applicable to 11, the
issue was a regression only applicable to 13 and above. Per your query
on the issue itself, the fix was not placed into certified 13.1 as no
Digium customer requested it to be.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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or unbound was too strict and has been
tweaked since the initial beta1 release. The next beta (or rc) will have
the fix, and it's confirmed to work against Ubuntu 14.04.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.
I'm having ??
WebRTC requires SRTP and Asterisk has to be built with it enabled. It's
okay if pjproject doesn't as we don't use their media layer. Do you have
the res_srtp module in Asterisk?
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - U
reasonable behavior.
[1] https://issues.asterisk.org/jira
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Joshua Colp
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Check us out at: www.digium.com & www.asterisk
up to the provider or gateway)
it's just answered instead.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
in.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
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r but some numbers will stay in
inband progress (unanswered) for a bit. Some toll-frees for example.
The specific SIP message that would show it as answered would be a 200
OK to the INVITE we sent though. If you provided the SIP log then we
could see.
Cheers,
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Joshua Colp
Digium, Inc. | Senio
/
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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it the source will always be it. That is if you do a SIP call to
Asterisk then media will come from that same Asterisk.
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Joshua Colp
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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/asterisk-users/2016-June/289326.html
[3] http://blogs.asterisk.org/2016/02/17/odbc_gutting/
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Joshua Colp
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
in it to override built-in defaults for
everything. You have to specify it yourself for realtime. If using
config files then config file templates can be used to do this.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out
doing the actual release!
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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not using res_odbc like it should be, which
causes issues as a result of a fixup from some multi-connection
problems. We'll get it sorted.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
you should create a new issue. If you don't want to
I can do so tomorrow. The complete console output (with debug going to
console in logger.conf and core set debug 3) as well as the
configuration would also be useful.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davi
. If another backtrace can be added we can see if it's the same thing.
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
rADeadlock
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Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com & www.asterisk.org
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thufir wrote:
On Wed, 29 Jun 2016 06:38:53 -0300, Joshua Colp wrote:
An INVITE is a request to set up a session, commonly referred to as a
call. Anything supporting SIP to establish calls uses INVITE to do so.
It's equivalent to picking up the phone and dialing a number.
an INVITE would
Andrew Ivins wrote:
On 1 July 2016 at 17:41, Joshua Colp <jc...@digium.com
<mailto:jc...@digium.com>> wrote:
exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
same => n,Dial(PJSIP/phone123, 30)
Your exten line has no priority,
n isolate things a bit further by trying the following:
Set(CALLERID(all)=Jon Doe <+123456789>)
Or individually:
Set(CALLERID(name)=Jon Doe)
Set(CALLERID(num)=+123456789)
If those don't work I'd suggest showing the console and your
configuration for the endpoint as it's something there i
to get thrown off in the environment.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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ng related or not.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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and extensions.conf is the logic which
describes when to place an outgoing call when we get an incoming call.
Whether you must use Asterisk is really up to your experience, what you
are trying to do, and whether other things can do what you need.
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Joshua Colp
Digium, Inc. | Senior Software Developer
Valter Nogueira wrote:
I am using Asterisk 13.9.1 and want to catch AttendedTransfer, but it is
not fired at all.
You'll need to describe more about the scenario. What exactly are you
doing, and where are you expecting the event to appear?
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Joshua Colp
Digium, Inc. | Senior Software
+and+Mailbox+State+Using+PJSIP
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Joshua Colp
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Simon Hohberg wrote:
On 06/27/2016 12:09 PM, Joshua Colp wrote:
Simon Hohberg wrote:
Hi,
I want to pass a part of a SIP INVITE multipart body. I found a quite
old patch here:
https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22
But this patch
for the relevant code?
There is no option to block this and you would need to modify chan_pjsip.c
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out at: www.digium.com & www.asterisk
It appears as though something is requesting a channel but not giving
enough information. Are you using AMI or have something outside of
Asterisk that is getting information about things? What you'd need to
investigate is the console output and that information.
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Joshua Colp
Digium, Inc. | Senior
a patch? What do I have to put in the
dialplan then?
If you are asking if you can manipulate or get this information from the
dialplan in PJSIP it's not currently possible.
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Joshua Colp
Digium, Inc. | Senior Software Developer
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Check us out
, res_hep_rtcp.so, and res_hep_pjsip.so
modules loaded? Are you using PJSIP? Are there any warnings/errors at
startup about it?
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Joshua Colp
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Check us out at: www.digium.com & www.asterisk
, and the underlying sorcery
should allow it... but it's not something that has been tested.
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Joshua Colp
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Check us out at: www.digium.com & www.asterisk
sterisk 13? Is there a
document that shows steps to a successful migration?
You should always clear your modules directory to ensure there's no old
leftover things, besides that it should be fine. Were there any errors
output in the console when loading?
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Joshua Colp
Digium, Inc. | Senior Software Develop
what's up.
We now return you to your regularly scheduled database usage!
Cheers,
[1] https://gerrit.asterisk.org/#/c/2943/
[2] https://issues.asterisk.org/jira
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out
Marek Červenka wrote:
Dne 6.6.2016 v 17:42 Joshua Colp napsal(a):
Happy Monday all,
Since I sent my previous email a lot has been learnt about our
UnixODBC problem and a path has emerged ensuring both better
performance while
making sure people are not required to upgrade their UnixODBC unless
1] and any feedback would be welcome, both
from code review itself and testing.
Cheers,
[1] https://gerrit.asterisk.org/#/c/2943/
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & ww
!
Cheers,
[1] http://mailman.unixodbc.org/pipermail/unixodbc-dev/2016-June/001890.html
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Joshua Colp
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Check us out at: www.digium.com & www.asterisk
nt changes in how ODBC
support works (we gave more responsibility to UnixODBC for things)
though there have been some crashes and problems which are being
investigated.
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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full log may provide
further enlightenment.
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Joshua Colp
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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"Invalid crypto suite: %u\n", suite);
It could maybe be as simple as that, but until it is attempted it's
unknown what other stuff may need to be adjusted.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Chec
tem' is not reloadable, maintaining previous values
-- Remote UNIX connection
-- Remote UNIX connection disconnected
bkk*CLI>
Since you did not change the file it did not do the needless work of
rereading it, so you didn't see the message again.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan
/browse/ASTERISK-26007
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Joshua Colp
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also enforce limits to ensure that a call that was not
properly terminated does not result in excess charges.
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case even when fudged together.
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via the
"digit=#" option, is that correct?
That's correct I'm afraid. If you need even more, then without modifying
the code that option won't work for you.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Ch
to be achieved here.
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+ # was the only way to
let them select.
Right, and since the subject of this thread is "Music on Hold streams" I
assumed he was doing that through a music class for each stream.
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Digium, Inc. | Senior Software Developer
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Ch
.
Cheers,
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ody to help me.
This can be configured in voicemail.conf using the "maxsecs"
configuration option. I don't believe this is exposed using the
Voicemail application options, just using the config file.
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
Michael Ströder wrote:
Joshua Colp wrote:
Michael Ströder wrote:
HI!
I'm trying to compile asterisk 13.8.2+ on openSUSE Linux but it fails. It seems
file ./main/libasteriskssl.so.1 is present when it fails. Building 13.7.2 works
without any problem. It fails since 13.8.0.
$ ./bootstrap.sh
;menuselect/menuselect --enable chan_ooh323
$ make
..
failure (see message below)
Any hint is appreciated. Thanks in advance.
Does this problem still occur in 13.9.0-rc2?
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t;no" value. Since "never" was
the default, but most users probably expect "no" this patch updates the
default for the "progressinband" setting to "no."
This was tracked under ASTERISK-24835[1].
[1] https://issues.asterisk.org/jira/browse/ASTERISK-
causing it to
send out of band ringing instead.
>
> Is it possible to switch off the standard behavior of asterisk /
> progrssinband for ring groups only by setting some other options?
Asterisk does not have the concept of ring groups, this is a FreePBX
construct. Asterisk itself does all
ry in the
signaling.
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attempting to reproduce the above and get a backtrace[2]?
[1] https://issues.asterisk.org/jira
[2]
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
so you'd have to do something.
Other phones just allow it to be provisioned. If you did it outside of
channel creation it's still lots of work to coordinate things.
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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be problematic.
Cheers,
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barging on the Local.
This is due to a limitation in the underlying mechanism that does not
allow the position to be defined of where to insert the barging and
recording. It's fixed so recording comes first, followed by injecting media.
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Joshua Colp
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me part of pjproject (to be accepted by the
maintainers)?
Yes. Any patches are submitted upstream and if the next pjproject
release has them then they are removed from Asterisk.
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Joshua Colp
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you can configure things to work that's great but I don't see any code
changes we can do.
Cheers,
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Joshua Colp
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
Robert McGilvray wrote:
Hello,
This build fails to load res_pjsip.so, it kicks back symbol lookup
errors for ast_pjproject_get_buildopt. Certified cert4 works fine,
pjproject is 2.4.5.
Are you selectively loading modules? If so you need the new
res_pjproject.so loaded.
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Joshua Colp
provided
information about chan_sip on your -dev cross post.
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Check us out at: www.digium.com & www.asterisk
proxy URI.
Example:
sip:example.com;lr
If used in a configuration file:
sip:example.com\;lr
The '\' is so the configuration parser does not treat it as a comment.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out
them. As for
a repo to use - FreePBX bases their distro on CentOS so theirs might be
best for you.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
Carlos Chavez wrote:
On 4/5/16 3:17 PM, Joshua Colp wrote:
Carlos Chavez wrote:
I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
, and ConfBridge. If it's strictly
a two party call then Asterisk forwards media as received.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk
PBX ?
Yes. :D
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by
itiated using the trunk1 endpoint then the
From user will be 'trunk2'. If this is received by another chan_pjsip
it will attempt to look for an endpoint named 'trunk2'.
This will allow you to differentiate.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville
Harley Peters wrote:
What do I create the issue under Asterisk or something else?
Don't really see a category for it.
Asterisk is the project and you can leave the rest as-is. It'll get
changed if it makes sense.
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive
Harley Peters wrote:
Never done it before.
Where url?
The issue tracker is located at https://issues.asterisk.org/jira and an
account can be created using the site at https://signup.asterisk.org/
Cheers,
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
-sounds-en_AU-alaw-1.5.tar.gz] Error 8
make[1]: Leaving directory '/usr/src/asterisk/sounds'
Makefile:457: recipe for target 'datafiles' failed
make: *** [datafiles] Error 2
This issue should now be resolved and I've confirmed that the link now
works. Sorry for the inconvenience!
Cheers,
--
Joshua
?
Is NAT in use on both sides?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
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