off of the Contact. You can get the Contact via AMI by
listening for events and by querying for the status of the contacts
[1].
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_PJSIPShowRegistrationInboundContactStatuses
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive
or MixMonitor?
With what application arguments?
If you look at a packet capture, does the packet capture reveal
anything about the jitter, and on what call leg?
Have you tried using a JITTERBUFFER?
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Hun
n those messages are generated.
If that doesn't fix it, then you may have some form of malformed RTCP
packet that is causing Asterisk to think that it has a slew of SR/RR
reports to generate. You may want to look at the RTCP information in
wireshark to determine how many RR/SR reports are being gene
w to Asterisk? Start here:
>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.
ND=1))
>
> exten => example_839,23,NoOp(${PJSIP_AOR(example_240,contact)})
>
> exten => example_839,24,ExecIf($["${PJSIP_AOR(example_240,contact)
> }"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER} example_240))
>
> exten =>
henanigans and/or custom code - than a second instance of Asterisk
will understand and read that JSON just fine. Assuming it was told to get
that information from its AstDB via Sorcery as well.
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us ou
> On Jul 15, 2018, at 11:37 PM, Naftoli Gugenheim wrote:
>
> Crickets...
>
> I've tried this now on 15.5.0. Still completely broken.
>
>
I suspect you’re encountering behavior that is working as intended.
Normally, when Asterisk plays back a file, it scans the file system for all
files wi
;
> On Fri, Jul 20, 2018, 11:45 AM Matthew Jordan <mailto:mjor...@digium.com>> wrote:
>
>> On Jul 15, 2018, at 11:37 PM, Naftoli Gugenheim > <mailto:naftoli...@gmail.com>> wrote:
>>
>> Crickets...
>>
>> I've tried this now on 15.5.
#x27;local'
> [Jul 20 20:00:21] > 0x7f9e8000cb00 -- Strict RTP learning complete - Locking
> on source address 127.0.0.1:7078
> [Jul 20 20:00:21] -- AGI Script agi://127.0.0.1/route
> completed, returning -1
> [Jul 20 20:00:21] == MixMonitor close filestream (mixed)
> [Jul 20
at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users
On Wed, 29 Aug 2018 22:52:05 -0400,
> Matthew Jordan wrote:
> >
> > [1 ]
> > [1.1 ]
> > [1.2 ]
> > On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group
> wrote:
> >
> > Depending on log trolling (Asterisk security log) misses a lot, and
> also depe
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11! Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https:/
that matter, many of the monolithic dialplan applications have specific
options that place channels into dialplan contexts that execute after their
execution. I'm not even sure I can begin to wrap my head around what that
will do to a channel in ARI.
--
*Matthew Jordan*
Digium - A Sangoma Compan
P packets until it locks onto an RTP source. It does this
to prevent media injection attacks. The default probation period for an RTP
source is four packets - you can configure the probationary period as well
as whether or not strict RTP checking is enabled in rtp.conf.
Matt
--
Matthew Jo
o manipulate/retrieve information from, as opposed
to relying on the two-party nature of bridges.
This usually works pretty well, except for CDRs, which are generally a mess
no matter what. :-)
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806
ist [1], where I responded that I would get answers to the licensing
questions. Granted, it has been much longer than a week or two - mea culpa
on a bad time estimate.
[1]
http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/0001
ne in Asterisk 10.
So, to everyone who helped make Asterisk 10 successful, thank you!
Matt
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com &
lock of code
that you've written with no context and asking someone to debug the
problems you're seeing is unlikely to generate the help you want.
Thanks -
Matt
[1] http://lists.digium.com/mailman/listinfo/asterisk-dev
[2] http://lists.digium.com/pipermail/asterisk-dev/2014-January/0645
RTP source socket address. This option only comes
; into play while using strictrtp=yes. Consider changing this value
; if rtp packets are dropped from one or both ends after a call is
; connected. This option is set to 4 by default.
; probation=8
Matt
--
Matthew Jordan
Digium, Inc. | Engine
. As always, thank you all
for your continued support of the Asterisk project - and the Asterisk
community!
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
__
not try and please everyone and just defined CDRs for
how we thought they should work; the behaviour of CDRs in Asterisk 12
and in future versions should be substantially more predictable.
Matt
[1] https://wiki.asterisk.org/wiki/display/AST/New+in+12
[2] https://wiki.asterisk.org/wiki
zation.
It doesn't show up in the CLI due to the xmldoc API not parsing out
that attribute. The same is true for the wiki documentation; that
project is up on github [1]. It wouldn't be a large patch to either to
have that attribute displayed.
Matt
[1] https://github.com/asterisk/publi
ve the exact error message that pjproject gave when you ran into
this problem?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
___
the media to a separate thread; Monitor attempts to
record the audio on the thread servicing the channel(s).
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
so we need to know the exact messages. Alluding to
error messages without providing them usually leads to more confusion, not
less.
[1]
https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject#Installingpjproject-IssuesandWorkarounds
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
that dials the SIP channel, and use
SIPAddHeader from there. A quick Google indicates others have used a
similar approach in the past as well [1].
[1] http://lists.digium.com/pipermail/asterisk-users/2008-January/204375.html
Matt
--
Matthew Jordan
Digium, Inc. | Engi
g into the
parking bridge as it knows that you have not yet safely left the
bridge you are in.
We'll take a look and see if there's a way to allow this to happen
again. For now, you should use the one touch parking feature.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445
On Thu, Jan 30, 2014 at 5:48 PM, Justin Killen
wrote:
> After posting this, I ran across 'core channel show concise', which gives
> the data in a more machine friendly format.
>
>
That may work over AMI, but in general, it isn't recommended. The
command class authorization, EVENT_CLASS_COMMAND, i
On Mon, Feb 17, 2014 at 4:29 AM, Nick Cameo wrote:
> Hello Ishfaq,
>
> I just tried it and it did create a P-Asserted header however it
> contains the extension
> of the asterisk peer not what was passed by our switch. From the
> previous example:
>
> P-Asserted-Identity: "222" (which is asterisk
dialplan, it will
never get put into the 'h' extension, unless you use the Dial
application's 'e' option. If you want hangup logic and you're using
Asterisk 11+, you could also use a hangup handler on the outbound
channel.
But otherwise, I would expect that the '
the userfield is that, on two channels in a
bridge together, the userfields are concatenated together using a ';'
as a delimiter.
2) Use the MASTER_CHANNEL function to reach back to the parent channel
and set the CDR variable there.
Matt
--
Matthew Jor
27;d probably have something similar to PJSIP_ENDPOINT, such as
PJSIP_AOR or PJSIP_CONTACT (or something like that), that lets you get
at the run-time information of an AOR.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://di
t the INVITE request with a 488 (you didn't offer me DTMF!)
(2) Accept the INVITE request but not have DTMF over RFC 4733.
What you're seeing is option (2), which I think is better than
rejecting the entire call simply because the thing you are talking to
doesn't support the DTMF m
that is analogous to the
chan_sip 'auto' setting - what you configure for you endpoint today is
what it will use.
That's not a bug, just something not existing yet.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
C
ome custom code in
ConfBridge, namely with the application "MyConfbridgeCount":
static const char *const app2 ="MyConfbridgeCount";
You should contact the author of that code and ask them to fix the crash.
but since it
was generated against a much older version, it would have been
difficult to apply to 1.8.26.0.
I've updated the patch on the downloads site such that it is now a
patch against 1.8.26.0. Let me know if you have any other issues.
Thanks -
Matt
--
Matthew Jordan
Digium, Inc. | E
d out,
there are still plenty of ways to manipulate CDRs through the
dialplan.
A specification for CDR behaviour in Asterisk 12 is available on the
Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Dav
k, specifically Section 1 of the GPL (if
you distribute the modified source in any fashion) and/or Section 2c.
Unless you really know what you're doing with regards to software
licensing, I would highly suggest not modifying the welcome message.
--
Matthew Jordan
Digium, Inc. | Engineering
ience with it. We do, however, use
starpy (https://github.com/asterisk/starpy) extensively in the
Asterisk Test Suite. It does lock you into using twisted
(https://twistedmatrix.com/trac/) - which has both pros and cons - but
it may be a viable alternative for you if pyst doesn't work out.
Matt
-
; in this situation, Alembic is
far more useful.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
e=user
; The name is the text between square brackets [name]
; 2. Asterisk checks the From: address and matches the list of devices
; with a type=peer
; 3. Asterisk checks the IP address (and port number) that the INVITE
; was sent from and matches against an
gt; did the trick, but the install-prereq script wasn't good enough.
>
What distro are you building on?
I'm running both Ubuntu 12.04 and CentOS 6.5 locally. Both have the
libraries listed in install_prereq.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive
sions 0.x, every minor version is assumed to be
incompatible with every other minor version.
{quote}
http://www.webdav.org/neon/doc/html/refvers.html
You should either downgrade to 0.29, or else have a community
developer determine if res_calendar_ews is compatible with later
versions of neon.
Mat
l be in 'host' property. I assigned as
> host=[IPV6]...but it shows error.
> Can anyone help with this issue.
>
>
IAX2 does not support IPv6 in that version of Asterisk. IPv6 support was
added to chan_iax2 in Asterisk 12 [1].
[1] https://wiki.asterisk.org/wiki/display/AST/New+in+1
ging easier just recently). Enable REF_DEBUG in
menuselect under Compiler Flags, make/make install, and re-run the scenario
that reproduces the result. A refs file will be created in your Asterisk
log directory - attach that to the issue along with DEBUG log.
Thanks!
--
Matthew Jordan
Digium, In
be a bug, the original issue will get re-opened.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Band
e's a ref leak (since otherwise,
the CBAnn channel would be long gone). If you can get a ref debug log and
the standard Asterisk DEBUG log showing the problem, that would help a lot
in finding out what is going on.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 J
the refs log and the full
DEBUG log. That will allow us to understand what's occurring here.
Thanks -
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
___
s part of it.
>
> Is it unwise to use channel names to extract the peers involved in a call?
>
>
>
How a channel is named is a function of the channel technology. Which
channel technology(ies) are you curious about?
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan
, unfortunately not.
It would be a relatively trivial addition to add a dialplan application
that could emit an Asterisk logging message at any one of the various
levels, if someone were interested.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL
On Wed, Apr 30, 2014 at 8:37 AM, Administrator TOOTAI wrote:
> Le 30/04/2014 15:19, Matthew Jordan a écrit :
>
>
>> On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI
>> > ad...@tootai.net>> wrote:
>>
>> Please, people from Digium, Matt again
l,message)
>>
>>
>>
>> [Arguments]
>>
>> level
>>
>> Level must be one of 'ERROR', 'WARNING', 'NOTICE', 'DEBUG', 'VERBOSE'
>>
>>
k.org/wiki/display/AST/Asterisk+11+Function_DB_KEYS
* https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_DBdeltree
Whether or not you store 'contact information' (and that could have a
variety of meanings, so I won't interpret it specifically) is up to
you.
--
Matthe
started. If your log doesn't show that, then
there may be another generator present that is preventing silence from
kicking off.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
ain at a confirmed state if a
> second call came in while already on a call.
Unfortunately, notifyringing is only set in the [general] section in
sip.conf. It does not have a peer level override.
It would be nice if it was set on a peer by peer basi
de later
to escalate it to 'core stop now', by default, Asterisk will refuse
the 'core stop now' command. You can, however, stop the 'core stop
gracefully' by issuing 'core abort shutdown', which will cause
Asterisk to stop the existing shutdown attempt and r
On Wed, May 28, 2014 at 1:08 PM, Matthew Jordan wrote:
> On Wed, May 28, 2014 at 12:47 PM, Doug Lytle wrote:
>>>> Perhaps i should join the -dev list to find out what 'convenient'
>>>> actually means for the process...
>>
>> The dev list is
- can use TLS as a
transport. If your OpenSSL version is one of those affected by the various
vulnerabilities, then yes, you are at risk.
This also applies to all other modules in Asterisk that use TLS, including
AMI, the HTTP server, and others.
Matt
--
Matthew Jordan
Digium, Inc. | Engineeri
was to remove the visibility of masquerades from
external systems (and mostly purge them internally), such that channels
have a stable, consistent identifier for the channel throughout its
lifetime.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 3580
ialplan for the outbound extension, you dial yet another
Local channel. I would expect this to result in 3 CDR entries:
Source Channel Destination Channel
Local/queue@TiagoGeada;2
Local/queue@TiagoGeada;1 Local/932485427@outbound;1
Local/932485457@outbound;2
So, the que
>
>
There was a bug in secure WebSockets (tracked under ASTERISK-21930) that
was fixed in Asterisk 11.9.0:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.9.0-summary.html
Which version of Asterisk are you using? Is it 11.9.0 or later?
--
Matthew Jordan
Digium, I
negotiation of DTLS and
Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
ttps://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_CHANNEL
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
__
ideas where to start
> looking for the problem?
>
Please get a backtrace illustrating the problem:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
Once you have a properly generated backtrace, open an issue on
issues.asterisk.org.
Thanks -
Matt
--
Matthew Jordan
Digiu
RTP is merely swapped between ports) and a remote bridge. The remote
bridge is where the two channels are in a bridge in Asterisk, but
media flows directly between the endpoints.
If your endpoints are behind a NAT, then no, you cannot use a remote
bridge. No amount of hoping or tinkering will mak
-INVITE).
; Additionally this option does not
disable all reINVITE operations.
; It only controls Asterisk generating
reINVITEs for the specific
; purpose of setting up a direct media
path. If a reINVITE is
a, then debug your
network to determine why media could not be sent directly between
those two devices.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com &am
;
> Should I open a bug or there is something I am missing?
>
I suspect you have some configuration error or environment problem.
The Asterisk Test Suite - which runs on every commit and nightly -
makes extensive use of custom asterisk.conf files to sandbox instances
of Asterisk that run con
which
is a core supported module in Asterisk 12.
[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com &
res_pjsip_* modules couldn't be loaded
on a particular instance of Asterisk would be helpful.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: htt
yet,
unfortunately, was overlooked.
Ideally, it'd be in the CHANNEL function.
If anyone is curious, the accessor function you want is
ast_channel_callid. It returns the callid ref bumped, so you do have
to make sure you decrement the ref count using ast_callid_unref. You
can print the callid to t
showing XXX for pjlib
>
> Please let me know if any more information is needed
>
>
What is the output of "pkg-config --print-provides libpjproject"? For that
matter, does "pkg-config --list-all" show libpjproject as a package?
--
Matthew Jordan
Digium, Inc. | Eng
ons are mutually exclusive because GCC places a
> trampoline on the stack.
>
> The lack of NX-Stacks could be a security defect and could lead to
> governance problems.
>
I'm sorry you don't like nested functions.
The use of RAII_VAR has saved the Asterisk project o
ones
are preferred:
https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
For those who want the cliff's notes version, it is:
* res_timing_timerfd
* res_timing_kqueue (where available)
* res_timing_dahdi
* res_timing_pthread
In particular, res_timing_pthread should only be used as a l
is seems quite odd.
Keep in mind that asking for help with deployment issues on asterisk-users
is entirely appropriate, but do remember this is an open source project and
everyone who replies on here is doing so of their own volition. No one is
required to solve your issue for you.
--
Matthew
t the documentation fixed.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Bandwidth and Colocation Pro
with Oracle
use LENGTH, not LEN.
Your solution, as it is currently, wouldn't be acceptable, as it would
cause far more problems than it would solve. About the only way I
could see solving this would be to make it configurable some place.
Given the relatively few number of people who u
we're thrilled with that 5 second wait
time. See https://issues.asterisk.org/jira/browse/ASTERISK-23259 for a
bug report noting this behaviour.
> Why are you attempting to request an agent that has a device state
> (Agent:agent_id) of busy anyway? That agent could be on another call
On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik wrote:
> Hi
>
> Is anyone using asterisk on CentOS 7?
>
> If so, is it working fine and as expected?
>
Random data point: the Asterisk project's build agents are still on CentOS 6.
Your mileage may vary.
--
Mat
nk some more
technical details about Bleep would be helpful for the Asterisk
developers, so we could see what would be needed for Asterisk to
communicate with Bleep.
Thanks!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us o
edia.so) ?
>
Does pkg-config find libpjproject?
$ pkg-config --list-all | grep libpjproject
Asterisk's configure script uses pkg-config - so if that can't find
it, Asterisk can't find it.
--
Matthew Jordan
Digium, Inc. | Engineering Manag
1374 would be the port.
>>
>> /Mikael Fredin
>
> Sure but what I'm looking for is to:
> - type something like "rtp show settings"
> - and read something like : Port range 1-2
That information is not available via a CLI command.
--
Matthew Jo
ch a log are on the wiki [3]
* Execute the CLI command 'core show fd'. This will dump out all
allocated file descriptors. Attach the output of the command to the
issue as well
[1] http://lists.digium.com/mailman/listinfo/asterisk-biz
[2] https://issues.asterisk.org/jira/browse/ASTERISK
that aren't delivered with Asterisk?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Bandwidth and Co
bug. When things got ported over to hit the cached
snapshots of the channels (as opposed to locking the live channel),
that field got missed.
Please file a bug on issues.asterisk.org.
Thanks!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL
es.asterisk.org/jira/browse/ASTERISK-24234
You may want to try the patch on the issue to see if it resolves your
crash. Alternatively, you could try checking out the 12 branch.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
people may have had regarding Asterisk and the
UniMRCP project.
Thanks -
Matt
[1] http://www.unimrcp.org/
[2] http://www.gnu.org/licenses/license-list.html
[3] http://svn.asterisk.org/svn/asterisk/trunk/LICENSE
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsvill
wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
If you can reproduce the issue, that will help a lot as well.
Thanks -
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com
ns you may have
at members of the Asterisk Development Team (myself included).
More information about the hackathon can be found on the ChallengePost page
or at http://www.asterisk.org/community/astricon-user-conference/hackathon
See everyone in Las Vegas!
Matt
--
Matthew Jordan
Digium, Inc
and my sip.conf:
>
> [general]
> bindport = 5070
> bindaddr = PU.BL.IC.IP
> udpbindaddr = PU.BL.IC.IP
> tcpenable = yes
> limitonpeers = yes
> rtcachefriends = no
> tos_sip=cs3
> tos_audio=ef
> realm = testers.com
> autodomain=yes
> domain=PU.BL.IC.IP
> dom
gt; m=audio 18366 RTP/SAVPF 0 3 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=connection:new
> a=setup:actpass
> a=fingerprint:SHA-256
> CE:EE:D9:28:EA:B0:6E:D0
.9.1).
>
> https://github.com/fail2ban/fail2ban/pulls
>
> HTH,
> Patrick
>
Why would you not use the SECURITY log format, which have the exact same
format between chan_sip and chan_pjsip, and have a consistent format from
Asterisk 10+?
https://wiki.asterisk.org/wiki/display/A
* much in new versions...)
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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_
-- Bandwidth and Colocation Provided by
T/Asterisk+Versions
The success of Asterisk 1.8 is due to the involvement and support of
the Asterisk community. As always, thank you for your support of
Asterisk!
Matt
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://
e hood and things
managed to remain the same was the goal.
chan_pjsip does use a different set of rules for how it offers its
codecs, and should generally follow what it outlined on that wiki
page.
Matt
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsvill
work on whatever channel it was set on. If you are
going to use it on an outbound channel, then you should use a pre-dial
handler to apply it to that channel.
Matt
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
can't reproduce this. We've been running a lot of
tests with a variety of SIP clients over the past week here at SIPit -
both with and without ICE - and I haven't had a single instance of
Asterisk failing to provide any ICE candidates when it is properly
configured to do so.
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Matthew J
re using a flexible backend (such as
cdr_custom or cdr_adaptive_odbc), you can add a custom column to your
CDR records - such as 'clid_original' - and use the CDR function to
set that value prior to changing the caller ID:
exten => Set(CDR(clid_original)=${CALLERID(num)})
exten =>
Command_get+variable
[2] https://wiki.asterisk.org/wiki/display/AST/AGICommand_set+variable
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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people use it, other than it comes up from time to
time in the issue tracker (which is about the extent of my visibility
for usage).
>> 3) If DUNDi is not really used in modern set-ups, then what are my
>> alternatives?
>>
>> I really have searched and read and Googled everything
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