.in". It appears as
though the ACKs were handled appropriately. If there is something else then
you'd need to be more specific about what is happening.
--
Joshua Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
You also need to look at the ICE
candidates to understand the possible paths. As well this message really
belongs on the asterisk-users list or the community forums[1].
[1] https://community.asterisk.org/
--
Joshua Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive
On Thu, Sep 13, 2018, at 7:16 PM, Joshua Colp wrote:
> On Thu, Sep 13, 2018, at 7:00 PM, Matt Fredrickson wrote:
>
>
>
> > > I have two potential fixes (and two that aren't practical options I
> > > don't think but might be with knowledge I do
en migrated and is on Matt Fredrickson's list to look at. Even
before now it was not updated in quite a few years. If you need it immediately
then it can be locally generated using the "make progdocs" target.
--
Joshua Colp
Digium - A Sangoma Company | Senior Software Developer
445
f traffic from a source other than where it
> > would route back to, basically forcing an IP match based on route-based
> > address selection.
>
> Multiple IPv4 address are not very common among non-carriers.
>
> > Joshua suggested that before coding on this is started all
ing ContactStatus or changing it
substantially directly, the fact that I'm asking my original question is just
because I'd like to remove that code which consumes extra resources for
something which I personally think is redundant.
--
Joshua Colp
Digium, Inc. | Senior Software Devel
the
> plug, no events are generated.
If properly configured (qualify is enabled) then an event will be generated
that it has become unreachable[1]. Without qualify there is no way for Asterisk
to know this.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_ContactStatus
on this behavior?
Cheers,
[1] https://gerrit.asterisk.org/#/c/asterisk/+/9822/
--
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
hen they will be doing it,
though.
--
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 Colocat
On Thu, Jul 26, 2018, at 5:29 AM, Floimair Florian wrote:
> I did a quick search, but so far haven't found anything aside from
> triggering notify via Asterisk CLI.
>
>
>
> So my question is: Is there any way to trigger a NOTIFY via ARI?
There is no mechanism to do t
At a minimum the latest version that it impacts, but the version where it was
introduced can additionally be mentioned.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com &
P message I'd like to see go
> back to the
> trunk provider. With my patch, it goes out. Without it, nada.
>
> But... incoming calls can come from either of two IP's, and "sip set debug
> ip xx"
> can only monitor one. So, what I thought was happening may
indicate that
conditions have changed, which results in the bridge re-evaluating the
situation and potentially changing the underlying technology. As Monitor is
quite old it's entirely possible that it doesn't have the call to do that.
The code is the following:
if (ast_channel_is
see if anything stands out?
There is no cache or memory mechanism built into sorcery that is always there,
so I'm not sure where it'd have two objects stored or where to look. The "pjsip
show endpoints" command just asks the sorcery backends to return the data they
have, for e
> contacts" in the CLI.
> This also sporadically leads to calls being refused by Asterisk even
> though the called endpoint registered successfully and is idle.
I should also add - are you creating the contacts, or are you referring to
contacts as a result of inbound registra
uld need a little guidance.
I don't know if anyone has tested the use of ARI push configuration with a
database backed realtime, albeit there is nothing to say it shouldn't work.
What do you see in the actual database?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan D
> Thanks!
To Asterisk it's just another video stream, there's nothing different within
the SDP itself. The only change that was done was to trigger another
renegotiation when a stream changed within an SFU conference which happens
automatically. The rest of the work is all client s
as well
as low level Firefox debug[1]. As the browsers are a black box problems may not
end up appearing on the Asterisk side, or even the Javascript console leaving
you to guess what is going on (and sometimes guessing wrongly).
[1] https://gist.github.com/ibc/3a10b27812d99c8abd1b
--
Josh
d plan format. In the case of Chrome
this has to be translated to/from plan B to work.
Have you examined the Javascript console to see if the SDP is complained about
at all?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW
n written to not download the file if it hasn't changed
which would explain the behavior you are seeing. I think it'd be reasonable to
add support for the scenario and shouldn't be too bad depending on the code.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Dav
On Sun, Jan 28, 2018, at 7:59 AM, Joshua Colp wrote:
> On Sun, Jan 28, 2018, at 5:00 AM, Alexander Traud wrote:
> > In Jira, when I create a new issue, the fields Summary and Description
> > are marked with a red asterisk. I guess, that indicates a required
> > field. How
cross the various places about it and the quality
of issue reports has actually gone up as a result (generally the only back and
forth now seems to be to get backtraces, config information, or environment
information).
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis
13.66*22* already:
> - <http://www.freepbx.org/downloads/>
> - <http://wiki.freepbx.org/display/PPS/10.13.66+Release+Notes>.
I can't speak on this, but Matt can probably provide input on things next week.
--
Joshua Colp
Dig
ink it's a regression you can say so on
the issue, otherwise it's up to the person doing the triage whether they think
it's a regression or not based on the information provided. If it's iffy then
it can be discussed on the mailing list.
--
Joshua Colp
Digium, Inc. | Senio
really support. We accept
contributions and changes which help to make Asterisk work there but we don't
test on it and can end up accidentally breaking it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Chec
s own outgoing subscription and act on
NOTIFY requests, there is currently no API or support for doing such a thing.
It only supports receiving a subscribe and acting on it.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806
On Wed, Jan 3, 2018, at 3:53 PM, Richard Mudgett wrote:
> On Wed, Jan 3, 2018 at 5:08 AM, Joshua Colp wrote:
>
> > On Wed, Jan 3, 2018, at 12:03 AM, Richard Mudgett wrote:
> > > On Tue, Jan 2, 2018 at 5:41 PM, Joshua Colp wrote:
> > >
> > > > On Tue
On Wed, Jan 3, 2018, at 12:03 AM, Richard Mudgett wrote:
> On Tue, Jan 2, 2018 at 5:41 PM, Joshua Colp wrote:
>
> > On Tue, Jan 2, 2018, at 7:14 PM, Richard Mudgett wrote:
> > > The patch for https://issues.asterisk.org/jira/browse/ASTERISK-27206
> >
s that the option itself has evolved and
hasn't been clearly defined with a strict purpose. I don't think now is the
right time to solve that particular problem. If there is a defined use case
which is currently broken by the chang
the interface provides, but the
benefit being it can be used by anything that uses the interface. Is it
worth it? Could be as Corey mentioned.
If your goal, however, is to provide all the knobs and possibilities
that redis can do - then it obviously wouldn't be a good fit.
--
Joshua Colp
rcery + other features). Which one should be used
really depends on the use case. These sit on top of the config parser
and map things into C structures automatically and atomically. You
define how a category maps to a C structure (for example requires a
typ
ot be possible to shut them down with ARI.
The DELETE operation in ARI on a channel works on any channel, including
those not in ARI.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 -
d?
13 does not have push so I don't think it would really have an impact
there or need to be backported.
--
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
--
__
On Wed, Nov 8, 2017, at 02:43 PM, Michael Maier wrote:
> On 11/08/2017 at 06:14 PM Joshua Colp wrote:
> > On Wed, Nov 8, 2017, at 12:57 PM, Michael Maier wrote:
> >> Hello,
> >>
> >> since asterisk 13.18.0, I'm facing frequent disconnects like these each
&
antial
change was in July and before that November 24, 2015.
--
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
--
_
: default
> force_rport : true
> ice_support: false
> rewrite_contact: false
> rtp_symmetric : false
> send_pai : false
> send_rpid
stion, is there a "safe" asterisk method of doing a "sprintf"?
There is no Asterisk wrapper for it, but snprintf should be used instead
generally.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check u
en Digium employees do not
> do that. That guideline is still in effect, isn't it?
Technically yes but it's a comparatively small thing in the grand scheme
of things so it doesn't see much enforcement.
[1] https://gerr
_mwi_external.c
It's highly recommended to use one of them instead of rolling your own
configuration parsing. They take care of things (such as safe atomic
reloads).
The "safe string copy" you are thinking of is probably ast_copy_string.
--
Joshua Colp
Digium, Inc. | Senior Software
ki.asterisk.org/wiki/display/AST/Asterisk+Versions
>
> Best wishes, and sorry again about the confusion on this.
Kia ora all,
This is just a reminder that changes should no longer be cherry picked
into the 14 branch. Right now the supported branches are 13, 15, and
master.
Cheers,
--
Joshua Co
ttps://wiki.asterisk.org/wiki/display/AST/AstriDevCon
--
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
--
_
-- Bandw
een more precise:
>
> Asterisk Project Security Advisory - AST-2017-008
RC1 does not include the fixes, as RC1 was released before the security
advisory. The change itself, though, has been put into the 15.0 branch
and will go into the release.
--
Joshua Colp
Digium, Inc. | Senior Software
ave the option to select your AstDB backend.
It was chosen not just because of that but also because to a user
there's nothing they need to do to set it up or have it work - you just
run Asterisk and off it goes.
I'm more curious over why they didn't use func_odbc instead but chose
g list and start a discussion.
Cheers,
[1]
https://wiki.asterisk.org/wiki/display/AST/Stream+Support+Across+Asterisk
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & ww
. They'd also increase the size of frames, which is something
else to consider.
I've heard other people doing this outside of Asterisk instead, to avoid
having to do any modifications.
--
Joshua Colp
Digium, Inc. | Senior
ng versions. This can cause problems where
the configure script detects one version, but the paths of the system
result in the other version being used causing it to improperly use
things. If you look on the system for "libpjsip.so" and it exists in
multiple places, then that's like
nly in production environment, but it is possible
> to add some logging if needed...
I think in order to determine the proper path forward we need to
understand the specific scenario, the threads involved, and the specific
interaction that caused it to happen. For example I wouldn't expect the
sip_transaction from
> being deleted? Or is there a way to determine that transaction was
> deleted (by some callback)?
Have you determined what callbacks will occur that can cause it to
happen? Are you referring to timer entries for example?
--
Joshua Colp
Digium, Inc. | Senior
c+Opus
I've updated the wiki page with this.
>
> And there is a typo in the asterisk code source samples.
> configs/samples/codecs.conf.sample (max_bandwitdth to max_bandwidth)
Can you create an issue[1] for this?
[1] https://issues.asterisk.org/jira
--
Joshua Colp
Digium, Inc. | S
o add logic to determine when exactly to use it. It's
uncharted territory for the DTMF support in RTP in Asterisk.
--
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
--
___
the number of offered streams configurable by the user?
Ultimately it's up to the application that is handling the channel to
decide what it wants and that is decided in the moment, not ahead of
time based on configuration.
I think maximum and minimum are useful for enforcing some constraints
tho
. In this case what have you set for "endbeforehexten" in
the cdr.conf file?
--
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
--
___
ew backtraces on this and the linked issue. Would
> anyone be willing to take a look, my C skills are not that good?
The issue is in queue to be looked at by us (Digium). I have no
timeframe on that. Perhaps someone else in the community will take a
look.
--
Joshua Colp
Digium, Inc. | Senior Software
e NULL and why the format is no longer valid would uncover the
real fix.
--
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
--
_
answer.
I've created an issue[1] to track this but I do not know when it will
get worked.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-26991
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.
"Application can change the interval value of timers only on a
> global
> basis
> (perhaps even only during compilation)."
>
> Any hints as to where this code is located?
The code itself is in the transaction layer of PJSIP at
pjsip/src/pjsip/sip_transaction.c -
ed?
Early media doesn't change the channel state to AST_STATE_UP. This only
occurs if the channel is actually answered. I'd suggest providing more
details about the precise scenario including any dialplan.
--
Joshua Colp
Digium, Inc. | Senior Software Dev
e?
Yes, depending on the level of interaction we have it may be possible to
make things reflect reality better. We use PJSIP functionality to do
this REFER functionality so we have to see what it can do.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive N
to spec. The chan_sip implementation is
simpler and basic. This should indeed be filed as a bug[1] with
configuration and console output with SIP traffic.
[1] https://issues.asterisk.org/jira
--
Joshua Colp
Digium, Inc. | Senior Software Develope
cture (which holds the details of the Contact header from
the REGISTER) which is associated with the AOR. In fact I'd expect the
code to work already with that Aastra specific usage. When
res_pjsip_registrar got the unregister it would look for the associated
contact to remove, but as there is
is fine to me. CentOS 6 I don't really
have a comment on, I'm not in that ecosystem myself.
--
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
--
On Tue, Apr 4, 2017, at 12:23 PM, Joshua Colp wrote:
> On Tue, Apr 4, 2017, at 12:11 PM, Gaston Draque wrote:
> > These 13.13 links seem 404d
> >
> > [x]
> > http://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-13.13-curr
og-certified-13.13-current
> [x]
> http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert3
>
This should be resolved shortly once synchronization occurs. I'll keep
on top of it and post back once it is done.
--
Joshua Colp
Digium, Inc. | Senior
I file a new bug or is it enough to reopen
> https://issues.asterisk.org/jira/browse/ASTERISK-26735 ?
This would be under a new issue, as it's been that way since the
creation of IP based matching in PJSIP and isn't due to the SRV support
itself.
--
Joshua Colp
Digium, Inc. | S
e've had a line length rule I don't have any indication of it in
my text editor or count. I personally value readability above all. I
generally start a new line where it feels reasonable and flows best. I
don't think we need to enforce (or have) a line length rule and think
that people c
or should we first provide more information on the list?
Please file an issue on the issue tracker[1] and use codec_opus as the
component. We'll triage and ask for the needed information there.
[1] https://issues.asterisk.org/jira
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 J
erisk.org/jira/browse/ASTERISK-26445>
>
> While writing I have also had the following segfault:
>
> Should an issue be raised for this?
Asterisk shouldn't crash so any time there is one an issue should be
filed. The worst case is it is a duplicate and we link/close.
--
J
ame underlying problem.
[1]
https://issues.asterisk.org/jira/browse/ASTERISK-26718?jql=text%20~%20"ari"%20AND%20status%20!%3D%20closed
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check
.
I don't trust clients enough to expect that this behavior will work for
everything. If we can do the work ourselves in a way that guarantees it
works for all clients I usually opt for doing that.
--
Joshua
placed for
each channel drivers.
--
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 Pr
st_rtp_engine and opens the listening socket, or
> some guidance to at least identify the api calls to make that happen?
>
>
>
> I think I am close, but I am missing something.
The UnicastRTP channel driver should already allow the audio portion of
this. It puts the local address in
On Tue, Jan 31, 2017, at 10:38 AM, Joshua Colp wrote:
>
> I believe the problem is in Asterisk, not PJSIP. We have our own logic
> around the handling of sending outbound requests and in this case it is
> not working as it should. It may be a result of a slight behavior change
>
On Tue, Jan 31, 2017, at 01:29 PM, Michael Maier wrote:
> On 01/31/2017 at 05:15 PM Joshua Colp wrote:
> > On Tue, Jan 31, 2017, at 12:01 PM, Michael Maier wrote:
>
> [...]
>
> >>> We
> >>> don't pass the information to the other side, w
in some way, as changing codec
> originated by UA already works fine here.
Yes, we accept the incoming reinvite, negotiate it against the
configuration, update the formats on the channel, and change the
translation paths accordingly. If this results in no translation path
then the existin
m is in Asterisk, not PJSIP. We have our own logic
around the handling of sending outbound requests and in this case it is
not working as it should. It may be a result of a slight behavior change
in PJSIP, but I still think the fix is on our side.
--
Joshua Colp
Digium, Inc. | Senior Software Devel
he
request to do the reinvite could always be provided to the channel
driver and then it would decide what to do based on configuration.
[1] https://wiki.asterisk.org/wiki/display/AST/Stream+Support
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Dr
right?
Do any areas need further definition?
I look forward to all responses on this exciting subject! You can
continue discussion here or add as comments on the wiki page. I will be
monitoring both.
Cheers,
[1] https://wiki.asterisk.org/wiki/display/AST/Stream+Support
--
Joshua Colp
Digium, In
using Asterisk 13.11.2.
This shouldn't be happening... can you show an example?
--
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
--
___
anymore. However, if you
are expected to establish an outgoing connection to the remote side then
the logic that the connection has dropped and you can't reach them is
not true. The Contact in that case should be valid.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
already done.
--
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 http://www
gt; (https://issues.asterisk.org/jira/browse/ASTERISK-26686)
>
>
> The issue is happening daily on multiple servers at random times.
>
>
> Can anyone provide assistance on the cause so I can update the ticket
> with the correct information?
I have already triaged this issue a
x27;t itself trigger new
subscriptions or refreshes. It may "work" if we still send a NOTIFY
saying the subscription is terminated (which you're trying to avoid I
believe), the device might then start a new subscription... but we're
into client side behavior here.
--
depending on timeout value, I think
we need to understand precisely the scenarios where stuff is failing to
find the real solution. It may be that the original change just isn't
viable and needs to be reverted.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW -
ever useful to anybody so I'm
> leaning towards option 1 but we need feedback.
>
> This would be a change to 13, 14 and master.
Since it was not really useful I'm okay with 1.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis
ing bridge creation. Is it possible to
check in the bridge loop to see that binaural has been enabled but not
set up and set it up? This should not impact the mixing loop a large
amount during normal operation and overcomes the problem that the
settings_lock has right now. It would also allow AP
there a better solution to addressing this issue?
While I don't yet have an answer for this I've poked others and am
pondering options myself.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan
g changed
into account. In the case of the TCP/TLS code it's not something I'd see
outside code or developers using (the commercial modules certainly don't
use it) which is why I'm personally not opposed to accepting it.
Any other thoughts? Do we want to be strict and only allow on mast
t;
> Same on debian 8 compiles/links without problems
>
> Skipping --with-pjproject-bundled does make it compile on debian 7.
This was found already and fixed[1].
[1] https://gerrit.asterisk.org/#/c/4481/
--
Joshua C
d what you can do may be
incomplete.
--
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 an
ke I said there is no way to receive this information or to safely
query the information.
--
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
--
__
need to define what the queue
should do and then use the primitives to implement it.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ARI
[2]
https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Manipulating+Channel+State
[3]
https://wiki.asterisk.org/wiki/display/AST/Introductio
ly no mechanism to know what the other side when
answered has negotiated. It's been talked about previously that it would
be good to have such a thing, but it does not exist currently.
The only thing you can do is set your nativeformats to the negotiated
codecs and let Asterisk do the rest.
-
with or control the app_queue
application. It allows you to write your own replacement for app_queue.
ARI provides the primitives required to do so.
What were you looking to do from ARI?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Chec
d
be nice to know precisely the configuration and scenario that causes it
to want to delete something that doesn't exist. An issue would need to
be created with the configuration and logs with core debug of level 2.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davi
+1 to option 2.
--
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
esolve the issue.
They remain open, so noone that I am aware of is currently working on
them. When they are they get assigned and go to in progress.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us ou
omeone else has an idea on how to better do it.
Cheers,
--
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
--
_
-- Band
Thanks! I'll play a bit with codecs then as soon as I have some time. By
> the way, would you rather me open a jira or something like this for this
> effort, or is keeping on reporting here fine?
It will need a JIRA issue in the future regardless.
--
Joshua Colp
Digium, Inc. | Senior
date as it does have
> many
> knobs one can play with).
>
> As a side note, I anticipated the code is ugly in its current form, and I
> haven't checked if there can be leaks as it is. I've been away from
> Asterisk coding for a while, so fresh eyes can probably help spot t
nection is being closed.
>
>
> Can you confirm if Asterisk does send these packets on TLS transports?
The code itself does not distinguish (or know) that it is TCP or TLS. It
only knows that the transport is a reliable one and in that case does
the keep alive.
--
Joshua Colp
Digium, Inc. | S
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