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
> -- 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:/
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
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
#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
;
> 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.
> 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
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
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 =>
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.
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
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
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
nel_name(chan), filename,
ast_format_get_name(ast_channel_writeformat(chan)), preflang ?
preflang : "default");
Matt
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
__
> This seams easier, for the moment.
>
> I think I still need to better understand what are mixed Asterisk-Kamailio
> architectures main strengths
> compared to alternatives (Asterisk alone, Kamailio/RTPproxy, ...) but that
> is
re app_queue no longer scales for you,
you need to build your own queuing solution using Asterisk's APIs.
app_queue was not designed to scale across multiple Asterisk instances, nor
was it designed to scale up infinitely (which, of course, nothing is.)
Matt
--
Matthew Jordan
Digium,
were lost? Does this require
> RTCP compatible endpoint/phone, or something else?
>
>
The number of packets lost is determined based on RTCP information received
from the far endpoint. The number is accurate so long as Asterisk is
receiving RTCP information from the endpoint(s) in question.
M
On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
wrote:
> On 21-11-16 15:17, Matthew Jordan wrote:
>
>
> On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
> wrote:
>
>> Hello
>>
>> when using Asterisk version 13.12.2 I notice that it takes up to 30
>>
2
>
>
> I did not see this behaviour in previous Asterisk versions.
>
> Could this be a bug ?
>
>
There's not enough information here to know what is preventing the call
from occurring.
I'd look at a debug log between the caller entering the Queue and the
outbound call
uire the dahdi
kernel module to be installed and available. (I could be wrong on the need
for a physical card however, so your mileage may vary.)
- Upgrade to a version of the kernel that res_timing_timerfd supports.
That should be Linux 2.6.26 and glibc 2.8 or later.
Personally, if I were in your shoes, I'
ng with potential jitter)
- CPU utilization with an active conference (95% idle doesn't mean that
some core isn't pegged)
- Any potential transcoding issues or codec issues
Matt
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL
h and Colocation Provided by http://www.api-digital.com --
>
> 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 mailing list
r and the new callee
in the same bridge as the original callee.
This process could be repeated as many times as you want.
[1] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_HOLD_INTERCEPT
--
Matthew Jor
e), you can use the
AMI_CLIENT [1] dialplan function to retrieve the number of sessions
they have currently established.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_AMI_CLIENT
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at
easing them back to the dialplan.
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by
t;?
> Alternatively, how can dialplan check if there is any AMI user connected and
> decide dial plan execution?
>
> Thanks & Regards,
> Amit Patkar
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us
to manipulate the bridge themselves outside of
Asterisk's control (via attended transfers).
Matt
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
__
r one of the other
sound generation applications.
(3) Wait for one of your outbound channels to pass a 180 back, and
allow that to cause the inbound channel to ring.
[1] http://lists.digium.com/pipermail/asterisk-users/2016-August/289781.html
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davi
: No provider found,
> checking channel drivers for SIP - 111
> 49 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
> peer 111
> 50 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for
> SIP/111 - state 1 (Not in use)
> 51 DEBUG[-1]: d
ant. Note that you
need to use Originate instead of Dial, as you would otherwise have the
participant be bridged in a new bridge with whoever they dialed.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge
--
Matthew Jordan
Digium, Inc. | CTO
445
d on the wiki here:
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-Troubleshooting
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
lines for versions are available on the wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:
re-INVITE being sent to them, there's
something seriously wrong with that provider. This is pretty core
functionality in any SIP stack.
Matt
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville
.
Can you get a backtrace of the threads? [1] Make sure you have
DONT_OPTIMIZE and BETTER_BACKTRACES enabled. That should show us what
the threads are doing, which would give us a better idea of what is
spending all the time processing things.
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+B
sterisk's support of available ciphers both in DTLS and
SRTP.
Thanks Alexander!
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
__
started_guide
As a result, you will almost certainly need to solicit help from the
GOautodial folks. Things that are packaged up in such a fashion
typically have a specialized configuration that is too specific for
the Asterisk project itself to support.
Matt
--
Matthew Jordan
Digium, Inc. |
> because it is used to delimit a URI from a fragment identifier in URI
> references (Section 4). The percent character "%" is excluded because
> it is used for the encoding of escaped characters.
>
> delims = "<" | ">" | "#" | &quo
h to said dialplan
functions, and should implement their own stringent access control.
Matt
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
__
ttp://lists.digium.com/mailman/listinfo/asterisk-biz
While I know conversations tend to diverge sometimes, the asterisk-users
list should be about using Asterisk, and not about promoting some third
party service or software that may pertain to Asterisk.
--
Matthew Jordan
Digium, Inc. | Director o
55 ; In addition, you can specify a specific To: header by adding an
56 ; exclamation mark after the dial string, like
57 ;
58 ; SIP/sales@mysipproxy!sa...@edvina.net
While it might be nice if it didn't always use a scheme of 'sip', that'd
probably be categoriz
entially a replacement for it. cdr_odbc doesn't receive much
attention as a result.
Frankly, we should probably just remove cdr_odbc.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.or
t;:
"rtp_symmetric", "value": "yes"}, {"attribute": "context", "value":
"default" }, {"attribute": "auth", "value": "alice" }, {"attribute":
"aors", "value": &q
orefersub
> Server: Asterisk PBX 13.3.0-rc1
> Content-Length: 0
>
>
PJSIP is rejecting the inbound INVITE request as 100rel is required, but is
not in the Supported header of the inbound SIP INVITE request. I would
suspect that the UAC is doing things incorrectly by placing 100rel i
dge
>
> -- Executing [5082@dynamic-nway:5] Set("DAHDI/i1/6000-436",
> "DYNAMIC_FEATURES=") in new stack
> -- Executing [5082@dynamic-nway:6] MeetMe("DAHDI/i1/6000-436",
> "5082,1pdMXq") in new stack
> == Spawn extension (sipphonesconf, 6000
pt under a user without
sufficient permissions, and/or running/invoking the Asterisk CLI (via
"asterisk -rv") as a user with insufficient permissions.
I would double check:
(1) What user/groups own the various Asterisk directories (specified in
your asterisk.conf)
(2) What user/group you ar
' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
== Spawn extension (default, 1000, 2) exited non-zero on
'PJSIP/bob-'
-- Executing [h@default:1] NoOp("PJSIP/bob-", "") in new stack
-- Executing [h@default:2] Log("PJSIP/bob-
written the CEL entry. If that doesn't show up in the database, then
> it is either in the ODBC driver or the Maria database.
>
> If you don't see that message, then something is preventing those events
> from getting delivered inside of Asterisk, which would only occur if you
> had
. If that doesn't show up in the database, then
it is either in the ODBC driver or the Maria database.
If you don't see that message, then something is preventing those events
from getting delivered inside of Asterisk, which would only occur if yo
> outbound_proxy=sip:1.2.3.4\;transport=tcp
>
>
>
> REGISTER sip:1.2.3.4;transport=tcp SIP/2.0
>
> From: ;tag=f47f3ed2-0975-4ff0-bd3b-bd5c38e594c4
>
> To:
>
> Contact:
>
> Route:
>
> User-Agent: Asterisk PBX 13.6.0
>
>
In order to preserve the re
nge "from-internal" to
> "internal" in dahdi-channels.conf . So that just leaves the question of how
> this configuration ever worked at all.
>
>
Sounds like you may have hit step 6...
http://plasmasturm.org/log/6d
On Mon, Nov 30, 2015 at 11:34 AM, Ethy H. Brito
wrote:
> On Mon, 30 Nov 2015 09:40:50 -0600
> Matthew Jordan wrote:
>
> > On Sat, Nov 28, 2015 at 7:14 AM, Ethy H. Brito
> > wrote:
> >
> > >
> > > Hi
> > >
> > > I hav
nnel is executing within them. Terminating
an outer container of PBX flow without properly terminating an inner one
can inbalance the stack.
And just as a reminder, Macros are deprecated. They tend to have odd side
effects at times, and overly nesting Macros can result in a crash. You
should consi
not show up in configuration documentation. However,
since this is a sorcery object, you can specify in sorcery.conf where you'd
like that object to be persisted. Note that by default, it is persisted
using the 'memory' wizard.
--
Matthew Jordan
Digium, Inc. | Director of Technology
On Sun, Oct 18, 2015 at 12:39 PM, George Joseph
wrote:
> Did you open a Jira issue for this yet? I can actually work on this this
> week.
>
I think it'd be pretty cool.
George: want me to open an issue?
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan
d with an image.
You can send arbitrary text message to/from Asterisk using SIP MESSAGE
requests. The fact that the text is XML would be immaterial to
Asterisk. That's probably the closest way to send arbitrary data to
Asterisk without writing a specific new module in the PJSIP stack.
--
Matthew J
Can someone help me to solve my problem?
>
Do you have a g729 codec module loaded? If so, does it show a
translation path between g729 and gsm? If so, do you have sufficient
encoder/decoder licenses?
Matt
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan
On Sat, Oct 17, 2015 at 2:32 AM, Thyda ENG wrote:
> Can i send XML data over the asterisk PJSIP ?
>
That's a fairly generic question. Can you be more specific about what
you are trying to accomplish?
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW -
x27; and
'qualify_frequency'. Which one is currently giving the conversion
error?
Matt
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
ith anyone you might know that could spare $5 toward
> a good cause.
>
I'm pretty sure this has nothing to do with the Asterisk project.
Please don't e-mail this list again with non-Asterisk related
questions or topics. Doing so will get you kicked off of the list.
--
Matthew Jordan
it by specifying the User portion.
>
> However semi-colon is treated as a comment by the Asterisk parser. Adding
> quotes (“) around the setting doesn’t seem to help.
>
Use a '\', i.e.,
contact=sip:01234567\;tgrp=01234567\;trunkcontext=...
--
Matthew Jordan
Digium, Inc. | Direc
On Fri, Oct 9, 2015 at 8:27 AM, Ross Beer wrote:
> Hi Andrew,
>
> Unfortunately that has stopped working when using chan_pjsip and asterisk
> 13.
>
> The CDR is closed too early after a dial attempt. This is the expected
> behaviour for Asterisk 13, however you should be able to set the variable
>
3.6.0, and should be in the latest RC (13.6.0-rc2 [2]).
In either case, you're using a function as opposed to some
application, which means you do need to call the functions on the
specific channel. To get access to the outbound channel, you can use a
pre-dial handler's 'b' opti
your use cases. Would I like it to work well for you? Of
course! But if you don't participate by reporting issues, testing
changes, and contributing code, there's not much I can do for you
other than to note that the line is long, and feel free
On Mon, Sep 28, 2015 at 11:38 AM, Emil Ohlsson wrote:
> Ah, so I can use
>
> MessageSend(sip:alice)
>
> to send a message to Alice then (reusing the existing TLS session). That does
> seem to work. Thanks :-). I didn't know you could use users there.
>
> Is there a variable or some other method t
org/wiki/display/AST/Asterisk+10+Application_MessageSend
SendText is used for sending text messages within a call. Since a SIP
channel is not servicing the out of call text message, you cannot use
it to send a SIP MESSAGE request back to whatever sent the original
SIP MESSAGE request.
--
Matt
www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://
ot much anyone can tell you
unless they are familiar with that device. Asterisk is being told to
hang up the call, and so it will do so.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive
ertain requests while allowing others
through, then today, there is no way to accomplish that in the PJSIP
stack. As an open source project, someone could certainly propose that
functionality if they wanted.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville,
risk it uses TCP over a local network.
>
> I've been pulling my hair out for days. I really would appreciate any
> ideas or some pointing in the right direction here.
>
> Thanks in advance,
>
> C
>
> --
> _
lplans.
>
> Maybe this can assist someone else struggling with older 1.8 series timer
> issues.
>
> Regards
>
Always nice to hear that we fixed things. Thanks for the follow-up!
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806
this for Siren14 (I forget why).
>
Alas, until we get off our butts, yes. Sorry about that.
Really, we're putting as much effort into fixing things and issues
that affect a lot of people. While siren7/siren14/silk are nice, there
aren't as many people using them as other affected thi
CDR(userfield)=changed)
> exten => h,n,NoOp(${CDR(userfield)})
>
> In the same context I execute:
> exten => 10,1,Set(CDR(userfield)=empty)
> exten => 10,n,Dial(SIP/10)
>
> The "h" extension outputs two lines with userfield set to "empty". I would
>
> -11.19.0-rc1
> \ No newline at end of file
> +11.19.0
>
> \ No newline at end of file
>
> As you can see patch is build against 11.19.0-rc1, not 11.18.0
>
> How can we apply this patch to a legacy asterisk-11.18.0 tar.gz ?
>
> Thanks for any hint.
>
That's
0Hz AC hum present in
> some calls I push through my asterisk.
>
> Thanks
>
>
If you're willing to write C, then yes, what you're looking to do is
possible.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Che
g in the dialplan.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE
and https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE_DATA
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend
--
Matthew Jordan
Digium, Inc. | Di
rly install this package?
>
> I'm on CentOS 6.5 with updates, running on AMD Athlon +4400 and 64bit
>
> Thank you...
gmime is only required for the res_http_post module. If you don't need
that module, you really don't need that dependency.
--
Matthew Jordan
Digium, Inc. | Di
channel
with a PBX thread running. In this case, that's the two channels in
your output that are not outbound channels, i.e., the Local channels
that dialled your SIP channels.
That fact that you have two different SIP channels means that
something either performed two Originates, or you have don
https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
__
gt; connection to laf [laf]...
>
It looks like you are missing the outbound_proxy column on the
ps_contacts table. If you're missing that column, you are probably
missing some other columns as well.
Note that the schema for the realtime tables for PJSIP has been
updated many times, as n
media URI [1]).
For more information on ARI and its intended use, see [2].
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-play
[2] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573
Matt
--
Matthew Jordan
Digium, Inc. |
asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> _________
> -- Bandwidth and Colocation Provided by http://www
i. The basic tutorial example should give you an ARI event over a
WebSocket connection.
https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com
ve
made a note of it, and we'll keep evaluating it versus other planned and
requested features.
Thanks -
Matt
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at
happen at least once a day, however.
>
> What is the best way of getting the core show locks output for people to see
> as it appears to be too big to mail?
>
Please go ahead and make an issue on the issue tracker. Make sure you
get both the output of 'core show locks', as w
Can you provide a bit more information about the channels on
the PBX/Adhearsion server, who sends the REFER request, and what
happens explicitly in the scenario?
Matt
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: h
lifetime parameter in a crypto attribute is part of RFC 4568
(Security Descriptions for Media Streams), which Asterisk does not
support.
You will need to see if the Avaya system can be configured to not send
the attribute.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Dri
ence user usage patterns, we saw
MeetMe hit a limit at around 60 channels, and ConfBridge reach over
240 channels. Worst case for ConfBridge was around 140 channels.
Note that the ConfBridge sample rate, mixing interval, and other
parameters can greatly affect how far it scales out.
--
Matthew J
r supporting the Asterisk project!
Matt
[1] https://gerrit.asterisk.org
[2] https://git.asterisk.org
[3] http://lists.digium.com/mailman/listinfo/asterisk-dev
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.
t; (asterisk-current)
>Not found
>Available: pjproject-2.1-0.digium1.2_centos6.x86_64
> (asterisk-current)
>Not found
>Available: pjproject-2.3-5.el6.i686 (epel)
> Not found
>
>
> Does anyone have any id
gt; +-+++--++--+--+-+--+---+
> | 2015-03-26 12:11:04 | 1427382664.963 | 1427382664.963 | 645 |
> 5491549116 | 7051 | 27 | ANSWERED| SIP/pabx-e1-0
the crash will be needed as well. Instructions on
generating a backtrace can be found on the wiki here:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check u
-outisbusy, s, 5) exited non-zero on
> 'SIP/306-00b8' in macro 'outisbusy'
> == Spawn extension (from-internal, 0149XX, 7) exited non-zero on
> 'SIP/306-00b8'
> -- Executing [h@from-internal:1] Hangup("SIP/306-00b8"
asterisk is muggling the audio and video streams ?
>
> This is good information for all guys out there that wants to support video
> on webrtc in asterisk 13
>
Please stop spamming the list with this e-mail. Resending it multiple
times is clearly not yielding the results you'd like.
_threshold' and
'dsp_talking_threshold' settings:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk
On Thu, Mar 19, 2015 at 1:22 PM, Steve Murphy wrote:
>
> On Wed, Oct 29, 2014 at 7:10 PM, sean darcy wrote:
>>
>> On 10/29/2014 08:06 PM, Matthew Jordan wrote:
>>>
>>> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy wrote:
>>>>
>>>> Can
'native_rtp' basic-bridge
>
> -- AGI Script /pbx/agi.php completed, returning 4
>
Correct - and per the log, they shouldn't be in a direct media bridge:
> Locally RTP bridged 'PJSIP/99-0023' and
'PJSIP/304-0022' in stack
ng the two channels?
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Bandwidth and Colocation P
P capable channels.
Note that on verbosity 4, Asterisk will tell you if the bridge is
locally or remotely bridging the two channels.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us ou
provide what you're
looking for.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_message_context
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
eting the core files.
(3) The core files are hiding really, really well.
Either way, if you can't get a backtrace, there isn't much we can do
to help with that problem.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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