to figure
out what is occurring.
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issues.asterisk.org/jira/
[3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW
the transports to bind to port 5060, does that change anything?
--
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
to open an issue for it:
https://issues.asterisk.org/jira/browse/ASTERISK-24863
--
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
On Thu, Mar 12, 2015 at 5:11 PM, Chirag Desai djchill...@gmail.com wrote:
From: Matthew Jordan mjor...@digium.com
If the INVITE request is not shown in the CLI with 'pjsip set logger
on', then Asterisk is not actually receiving the request.
Does a pcap show the message being sent
On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes
steve-li...@geekinter.net wrote:
Anyone know where it’s gone?.. Appears to have been down all day.
The hamsters should be running in their wheels again now.
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW
? What
formats are negotiated on the channels? What symptoms do you see? What
does the CLI show, both when active calls are running and for a 'core
show channel' for the involved parties?
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
of invalid UTF-8 data from a Caller ID or Connected Line. When
you see this occur, what is the Caller ID/Connected Line name/number
on the involved channels?
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com
upstream provider is sending you, and chan_sip will
respond with a 200 OK.
--
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
thru sometimes.
How is the DTMF being transmitted from the phone to Asterisk? RFC2833,
in-band, SIP INFO...?
--
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
/Asterisk+Test+Suite+Documentation
[2]
https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suite
[3]
https://wiki.asterisk.org/wiki/display/AST/Running+the+Asterisk+Test+Suite
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
.
It runs just fine on Debian based systems. Most issues you will run into
are just making sure the dependencies are set up correctly.
It does require Python 2.6+ (recommended: Python 2.7 just in case something
has slipped in that we missed.)
--
Matthew Jordan
Digium, Inc. | Engineering Manager
trace the RTP
instance address (which is printed in that ERROR message) to what created
it.
--
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
()
same = n,Dial(PJSIP/Alice,,b(default^set_up_outbound^1))
...
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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
, and have it handle them accordingly.
Also, Kamailio itself has to be protected from failing, and probably even
from overload...
That's pretty standard stuff for Kamailio.
Would be great to read something in-depth about that
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis
of the context where the call is running in
that moment?
Set the TRANSFER_CONTEXT variable on the initiator of the transfer.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806
that, you do lose all of the location based
information. But that can be a good thing, if you tailor your applications
to expect that.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http
are you using?
--
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 Provided by http
interested in submitting the contribution upstream to the
project, you can find instructions on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
Thanks!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806
a drop in replacement for fail2ban.
-M-
P.S. My opinions are my own and do not necessarily represent those of my
employer. As an employee of Generation D System you can bet my opinions
are biased though!
It's nice to hear someone is making use of the AMI security events!
--
Matthew
Asterisk does not understand or support an SDP media type of 'message'.
Both chan_pjsip and chan_sip can support SIP MESSAGE requests, received
both in dialog and out of dialog. In addition, chan_sip will handle media
types of 'text' for real-time text received in the RTP stream.
--
Matthew
in advance for any info!
Features are only applicable while you are in a bridge. It certainly is not
a bug.
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
. If they are assigning
semantic meaning beyond constructing a globally unique identifier to
the IP address in the owner field, then they are taking a rather
extreme view of the RFC.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http
for me - but -
1)I don't know what else this may effect
2)I dont know how to pass this on to the development team
Please file a bug on issues.asterisk.org.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http
Asterisk 13! - this isn't a reason to go to Asterisk 13.
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
in contexts. Is there any application to finish processing the
extension in the context?
Which version of Asterisk are you using?
Can you provide a log showing the channel continuing on into the s extension?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW
.
There is no mechanism in Asterisk today to pass through a re-INVITE to
initiate a remote hold.
--
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
+Versions
Thank you for your continued support of 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
you'll need to provide some more information on how you're
producing this situation. Specifically:
* Channel technologies involved, and the formats on the channels
* Dialplan that reproduces the problem
Are you using any non-core dialplan applications or channel drivers?
--
Matthew Jordan
]. It exists
specifically for this purpose.
[1] https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Incomplete
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us
- but again, modifications occur on that CDR, not on previous
ones.
If you want the CDR for the channel prior to the 'h' extension to have
a userfield entry, you have to apply it before the channel hangs up.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL
how to debug/diagnose this?
Are the rest of the fields in your peers being extracted correctly?
Can you provide the output from your database for one of your peers?
Which realtime backend are you using?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL
:
Use Reason : No
Encryption : No
Which realtime backend are you using? (MySQL can be interfaced to with
a variety of backends)
--
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
your databases, and
use it to inform Asterisk of the failure.
--
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
/g.php?MTID=e1bc7fec6e11b4d0114accc8cf20b36e6
Flavio E. Goncalves
SipPulse
Please don't advertise on the asterisk-users mailing list.
Advertisements for products and services belong on the asterisk-biz
mailing list [1].
Matt
[1] http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Matthew
/AST/Getting+a+Backtrace
--
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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_
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is resolved, I'd highly
recommend looking at issuing a bug bounty [1], or contacting a
developer in the Asterisk Developer Community for assistance.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW
in the SDP. In the trace
that you provided, which request/response did you feel was in error?
--
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
are dialed at the same time.
The documentation page you reference should be updated to include that
detail.
How about this page instead:
https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW
] http://blog.codinghorror.com/what-if-we-could-weaponize-empathy/
--
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
and, preferably, BETTER_BACKTRACES
selected in menuselect.
Depending on the nature of the crash, you may be asked for more
information, but we won't know until we see the backtrace.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
and Channels page on the wiki has more on this here:
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Channels#IntroductiontoARIandChannels-ChannelsinaStasisApplication
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
some codec compatibility changes.
I would expect said modules to be available in the next few weeks.
--
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
behavior or an issue.
We saw in the documentation that the bridging module creates zombie
processes - is it related?
Where in the documentation (or in what documentation) does it say that?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
and
developer community does their best to test any new major version, but
obviously cannot cover every possible scenario.
Whether or not a new version is suitable for your configuration, in your
environment, is a decision you have to make.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445
/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Submittingthebugreport
It throws this error for me as well on the sample sip.conf, which does
have a udpbindaddr defined in the [general] context - so it's a
legitimate bug in the script.
--
Matthew Jordan
Digium, Inc. | Engineering
sometimes still occur in Asterisk 12+, they are far less frequent and
are no longer externally visible.
Why do you think you have zombie processes? Asterisk does use a large
number of threads, but generally rarely forks processes unless you are
using something like original AGI.
--
Matthew Jordan
://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation
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
and detect DTMF. Those
requirements are done by setting the various 'feature' flags on whatever
dialplan application is causing the channels to be bridged. For an example,
see the 't' or 'T' options in Dial:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
--
Matthew Jordan
Digium
)
--
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 Provided by http://www.api-digital.com
#Asterisk13RESTDataModels-ChannelVarset
--
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
? Are you
receiving other ARI events?
--
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
to occur.
[2] https://issues.asterisk.org/jira
--
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
, but instead simply looks at the column names and writes the
data values out to the file using the types that it expects each column
name to have.
So, changing those types will not work out well for you.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806
file a bug report ASAP at issues.asterisk.org, with a properly
generated backtrace:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http
with you in any fashion. Your tone, language, and
rhetoric are all indicative of someone who is not interested in having a
discussion or being a productive member of this open source community.
Good luck with your endeavors.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis
feel like they can bring up interesting,
radical, and yes - even crazy - ideas.
If you don't like that, you don't have to participate in the discussion.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com
On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht palbre...@glccom.com wrote:
On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote:
On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht palbre...@glccom.com
wrote:
On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote
On Fri, Oct 17, 2014 at 11:06 AM, A.Santoro n...@ecoricerche.it wrote:
On Wed, 15 Oct 2014 09:14:41 -0500, Matthew Jordan
mjor...@digium.com wrote:
On Wed, Oct 15, 2014 at 1:50 AM, A.Santoro n...@ecoricerche.it wrote:
Hi there,
I have installed Asterisk version 12.6 (on Debian wheezy
(answer)})
exten = ResetCDR(v)
Note the usage of 'v' to keep the custom variable that you just set.
--
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
puzzled that no one created a patch for the first timestamp when
a call is answered. If I get some free time, I will try to create one.
--
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
in MySQL record and in CVS.
Someone can confirm this event?
Without more information, there's no way to tell why that would occur.
Please provide a log showing the transfer with 'cdr set debug on' enabled.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL
:
Are you sure the channel is answered?
--
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
something, write a patch, and
submit it back to the project. [2]
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerEvent_Newstate
[2] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW
the result of 7 is what I
would expect.
--
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
] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
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
to find anything on configuring DUNDi with pjsip. Hoping one of you
people can point me in the right direction!
You may be the first person to try this out!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http
://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
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.
--
Matthew Jordan
Digium, Inc
(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 = Set(CALLERID(num)=6575309)
Matt
--
Matthew
are
going to use it on an outbound channel, then you should use a pre-dial
handler to apply it to that 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
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
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
+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
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com
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
--
_
-- Bandwidth and Colocation Provided by http://www.api
chan_sip and chan_pjsip, and have a consistent format from
Asterisk 10+?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http
=PU.BL.IC.IP:5060
I'd appreciate Your advice.
What does a DEBUG log show with 'sip set debug on' when the outbound call
is made?
--
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
:
-- Executing [661@default:2] Dial(SIP/660-0007,
SIP/661,3600,rt) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
What is their configuration?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http
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. | Engineering
://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 http://asterisk.org
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 - Huntsville, AL 35806 - USA
Check us out
://issues.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 us out
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 35806 - USA
Check us out
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
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_
-- Bandwidth and Colocation Provided
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-23721
[3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Matthew
On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik i...@pack-net.co.uk 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.
--
Matthew Jordan
Digium, Inc
to
communicate with Bleep.
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
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configure script uses pkg-config - so if that can't find
it, Asterisk can't find it.
--
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
something like rtp show settings
- and read something like : Port range 1-2
That information is not available via a CLI command.
--
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
://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 or in
a
between call wrap-up time.
--
Matthew Jordan
Digium, Inc. | Engineering
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
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_
-- Bandwidth and Colocation Provided
, 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 use MS SQL Server, I
wouldn't expect this issue to receive a lot of attention without a
patch.
--
Matthew
, this 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 Jordan
.
--
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 http://www.api
of RAII_VAR has saved the Asterisk project on countless
defects: memory leaks, reference leaks, port leaks, deadlocks on off
nominal paths, all sorts of ills.
They aren't going anywhere.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us
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. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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 the CHANNEL function's buffer using
ast_callid_strnprint.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan
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 http://asterisk.org
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: http://digium.com http://asterisk.org
instances
of Asterisk that run concurrently. This runs against every branch of
Asterisk and trunk.
I do not think there is anything wrong with the '-C' option.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com
, 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 http://asterisk.org
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 make it so.
--
Matthew Jordan
Digium
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