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
-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
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
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
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
--
__
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
--
>
>
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
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
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
- 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
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
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
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
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
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
l,message)
>>
>>
>>
>> [Arguments]
>>
>> level
>>
>> Level must be one of 'ERROR', 'WARNING', 'NOTICE', 'DEBUG', 'VERBOSE'
>>
>>
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
, 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
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
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
--
___
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
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
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
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
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
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
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
; 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
--
_
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
-
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
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
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
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.
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
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
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
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
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 '
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
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
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
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
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
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
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
--
___
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
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
. 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
--
__
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
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
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 &
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
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
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
more information on Asterisk versions and their supported lifetimes,
please see the following wiki page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Thank you for your continued support of Asterisk!
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW
Asterisk issue tracker - attach the log as well, as it may be useful in
analyzing how the system got into that state.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806
ine what module dependencies
need to be fulfilled. Beyond that, you should look at what applications and
functions a module provides to determine if you need it. Asterisk: The
Definitive Guide has some excellent information in Chapter 2.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Ja
ues.asterisk.org/jira
Make sure you mark the component as chan_ooh323.
Alexandr Anikin is the maintainer of chan_ooh323 [1]; he may be able to
correct the issue for you.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Open+Source+Maintainers
--
Matthew Jordan
Digium, Inc. | Engineering Manag
-AsteriskManagerInterface
*
https://wiki.asterisk.org/wiki/display/AST/New+in+10#Newin10-AsteriskManagerInterface
* https://wiki.asterisk.org/wiki/display/AST/New+in+11
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http:/
o pass a channel technology specific hangup
cause code is completely up to the channel driver. Not all channel drivers
support it; if someone wanted to add that functionality to chan_ooh323
that'd be great; but it's completely different than the condition that the
OP is seeing.
Matt
--
Mat
//wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12
[3] http://www.digip.org/jansson/
[4]
http://svn.asterisk.org/svn/asterisk/branches/12/contrib/scripts/install_prereq
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
44
d, arrive late, be incomplete or contain viruses.
> The sender does not accept any liability for any errors or omissions in the
> contents of this message which arise as a result of e-mail transmission. If
> verification is required, please request a hard-copy version.
>
> Please
re a wider audience may be available to assist you.
Thanks - and we all look forward to lots of productive discussions on the
new mailing list about building applications that use Asterisk as their
communications engine!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davi
o help you with this feature
request [2].
[1] http://svn.asterisk.org/svn/asterisk/branches/11/COPYING
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us ou
equirement for CDRs - ever - I
have to wonder, what is prompting this request?
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
--
__
#x27;d highly recommend using the version that is currently up for
review, as it will give you more bang for the buck. You can download the
patch from here:
https://reviewboard.asterisk.org/r/2846/diff/raw/
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Hun
DR(src) value ?
>
>
You can't. It is a read-only property.
If you want a custom value - "my-src" or something like that - you can add
a new value to your CDR record by using the CDR function, i.e.,
Set(CDR(my-src)=+34123456789). Certain CDR backends - such as cdr_custom
hone.so: undefined symbol:
> __ao2_container_alloc
>
>
Quite a lot, actually. Beyond just linking issues, there's that whole new
SIP stack thing we'd like to get it using.
We're working on it - stay tuned...
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Dav
l
you used was ARI, you'd lose some of the power of the dialplan. Likewise,
while you can do some of what ARI does via a combination of AMI/some AGI
variant, the result can be somewhat klunky and difficult to manage -
particularly for complex bridging scenarios.
Hope that helps!
Matt
--
Matth
n numerous bug fixes to the 1.8 and later
branches to address this kind of issue - since you're running a version of
Asterisk 1.8 that is 20 months old, there is a good likelihood that any
issue you are facing has already been fixed. Upgrading
t; list of the hangup numbers and the internal variable name look in
> include/asterisk/causes.h
>
> So if you change chan_sip.c and add the following just before the
> 'AST_CAUSE_NOTDEFINED' line and recompile and reinstall you should in
> theory be able to do a Hangup(44)
ew feature in Asterisk 10.
(Just in case someone is still running that version...)
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.
Use a '^' to delineate arguments pass to subroutines. This is actually true
for the U option as well. See:
https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
And:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
--
Matthew Jordan
Digium, Inc. | En
ouldn't receive any more e-mails from the Asterisk Wiki unless you
explicitly choose to watch a page.
Sorry for the spam!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com &
ation. However, if you use Local channel agents, then you could use
pre-dial to put the userfield information on the callee SIP channel when
the Local channel performs a Dial to the actual SIP device.
[1] https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
Matt
--
Matthew Jordan
agerAction_Originate
I would suggest setting that field to True in your Originate actions.
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
--
_
hronous Originate can
block that session from receiving events until it completes.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com
say it is "War and Peace", but yes, there is some content in
there.
--
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'll need some
more information to be able to tell what is going on. RTP debug for the
endpoint in question would most likely help.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out a
>
>
> Possibly, but not necessarily. Without seeing the whole backtrace it's
hard to say for certain.
The Asterisk wiki has instructions on how to properly get a backtrace from
a core dump created by Asterisk:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
Please do file
the transfer occurs?
You can also look at a trace of the SIP messages during the transfer using
'sip set debug on ' (set it for both the transferer as well as the
transfer destination). That should show why the requests are rejected
and/or why a call is hungup.
--
Matthew Jordan
Di
eing an American holiday and all, but we'll look into it ASAP.
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
--
__
me("SIP/1001-", "136512,,YBd") in new stack
[Jul 2 15:54:44] DEBUG[2737] app_playback.c: string
depth <0>
[Jul 2 15:54:44] DEBUG[2737] app_playback.c: try
in
The DEBUG statements in app_playback indicate the following:
* It will use the conf
g.
What specific problems are you seeing? What are the phones sending to
Asterisk, and what is Asterisk responding with?
A pastebin of a log showing DEBUG and higher level messages when a call
forward attempt occurs would help.
Thanks
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
4
you're operating on a level much closer to
what Asterisk is actually doing with its channels. This means having to
deal with Local channel pairs and - more importantly - masquerades. This is
a whole lot more powerful than CDRs, but does mean that you have to do some
bookkeeping to k
etc.) wants to
notify a channel that something has occurred, it queues a control frame on
that channel. Control frames include things like media source changes/media
updates, indications that signalling actions should take place, etc.
What specific use case are you looking at?
Matt
--
Ma
On Mon, Jul 1, 2013 at 6:24 AM, Amit Patkar | ATPL wrote:
> Hi
>
> I am using following say.conf file. Its a default file, which comes with
> Asterisk installation.
> When I call SAY DATETIME AGI function, it simply returns without playing
> date & time. Where as if I use mode=old setting, it wo
On Tue, Jun 25, 2013 at 12:18 PM, James B. Byrne wrote:
>
> On Tue, June 25, 2013 09:57, Matthew Jordan wrote:
> > On Mon, Jun 24, 2013 at 2:37 PM, James B. Byrne
> > wrote:
> >> It is not an infinite loop but it does go on for an inordinately
> >> long time.
It is not an infinite loop but it does go on for an inordinately long
> time. Does anyone here recognize what is happening and can provide me
> with an explanation?
>
Since it is pbx_spool doing the processing, you probabl
; Does this seem reasonable?
>
>
You can query a channel using the CHANNEL function (
https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL) to see if the
channel currently supports secure communication, and you can request that
the outbound channel be made secure using the same func
warded from one half to the other and vice versa.
Since you know that DAHDI/i1/96034296-30a3 is in a bridge with
Local/1004@from-queue-00019c34;1 and Local/1004@from-queue-00019c34;2* *is
in a bridge with IAX2/issuegroup-17175, you automatically know that
DAHDI/i1/96034296-30a3 and IAX2/issuegro
-b 2048
> -f 8192 Fermeture.mp3
>
>
> find /var/lib/asterisk/moh/
>
> /var/lib/asterisk/moh/Horaires/Fermeture.mp3
>
> ll
> -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010
> /var/lib/asterisk/moh/Horaires/Fermeture.mp3
>
>
>
>
>
Do you have the format_
telephone-event for ID 101
Capabilities: us - (alaw|h263p), peer -
audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Note that the port for the video stream is set to 0.
Asterisk is doing the correct thing: it notes that the answer to its offer
declined the video stream, so it disable
On 06/03/2013 01:03 PM, Chris Gentle wrote:
> On Mon, Jun 3, 2013 at 11:52 AM, Matthew Jordan wrote:
>> If both (1) and (2) are successful, than there's some impact that the
>> Ices application is having on the Local channel that is messing up the
>> reference coun
ended combination that *should* uniquely specify a
CDR (when configured correctly) is linkedid (which should be enabled and
added to your schema), uniqueid, and sequence number, with the asterisk
system name specified.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsvil
ually sinister)
If both (1) and (2) are successful, than there's some impact that the
Ices application is having on the Local channel that is messing up the
reference counting inside the ConfBridge. Otherwise, it's an error in
ConfBridge.
Matt
--
Matthew Jordan
Digium, Inc. |
variable that provides that level of granularity.
The closest available is the MEETMESECS channel variable, which tells
you how many seconds the participant was in the conference.
You can find a full list on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/MeetMe+Channel+Variables
Matt
terisk-1.8-current.tar.gz>
>
> May I now which one is the most suitable for a production environment ?
>
The Asterisk wiki describes the various versions of Asterisk:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
--
Matthew Jordan
Digium, Inc. | Engineering
hannel enters into an
AGI application using AsyncAGI, the same AMI connection can be used to
control the channel.
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
--
_
ay/AST/Asterisk+12+Projects
[3] http://lists.digium.com/mailman/listinfo/asterisk-dev
[4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asteri
201 - 300 of 420 matches
Mail list logo
