7; option
(2) Your core file may not be located in the directory that you are
running gdb from. You will need to find where the core file was
located - this is typically determined by
/proc/sys/kernel/core_pattern
--
Matthew Jor
On Thu, Mar 12, 2015 at 5:11 PM, Chirag Desai wrote:
>
>
>> From: Matthew Jordan
>>
>>
>> >> If the INVITE request is not shown in the CLI with 'pjsip set logger
>> >> on', then Asterisk is not actually receiving the request.
>> &g
ansport
> protocol=udp
> bind:0.0.0.0:5061
>
>
> [transport-tcp]
> type=transport
> protocol=tcp
> bind=0.0.0.0:5061
>
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 m
How the phones are negotiating media with Asterisk
Both your SIP configuration as well as a DEBUG log - generated with
trace logging, showing the negotiation [3] - will be needed to figure
out what is occurring.
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issue
IP endpoints.
Dmitriy was kind enough 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 u
On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes
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 - Huntsville, AL 35806
configurations? 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 - Huntsv
ut...
>>> Could you give me any idea why this error can appear?
>>>
>>> Thank you!
>>>
>>>
>> Well, looks like this is RTCP- because I get it every 5 second, which is
>> default.
>> Jus increased, to check is it RTCP or not...
>>
>
est URI. Make sure you have a matching
extension for what your 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
ill go 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://digiu
I have a CentOS VM
> specifically for running the test suite to avoid platform problems.
>
>
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
ionality than the unit tests.
[1]
https://wiki.asterisk.org/wiki/display/AST/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
ate DTLS
but is sending packets that are TLS encoded anyway, or it attempted to
negotiate but the UA wasn't configured for DTLS.
Generally, a debug log with 'rtp set debug on' will help you trace the RTP
instance address (which is printed in th
)=default)
same => n,Return()
exten => outbound_dial,1,NoOp()
same => n,Dial(PJSIP/Alice,,b(default^set_up_outbound^1))
...
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Hunts
, and
authentication, then you probably don't want Asterisk doing any of
that. You would instead just have Asterisk "trust" that Kamailio is
sending it the right calls, and have it handle them accordingly.
> Also, Kamailio itself has to be protected from failing, and probably even
>
d in the sip.conf instead 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
44
IP stack is very flexible, and can be made as smart or as simple as
you'd like it to be. If you want Asterisk to act as the registrar, than it
can do that and expose that information. It can also leave that job to
Kamailio - although when you do that, you do lose all of the location based
information
k 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 Provide
media->rtp) == AST_RTP_DTMF_MODE_INBAND))
> {
> return -1;
>
> That's it!!! It works fine for me. Any remarks / advice would be
> appreciated.
>
>
Hey Yaron:
Sounds like a good contribution. Since it is a code change, you may want to
discus
rules. It's 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 makin
tent-Length: 0
>
>
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 receiv
he presses a key while
> ringing in queue?
>
> thanks a lot 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 - Huntsv
concern, an arbitrary and private MAY be
chosen to populate the "o=" field, provided that these are selected
in a manner that does not affect the global uniqueness of the field.
{quote}
> So, I've got two questions here: First, how do I tell Asterisk / PJSIP which
> IP
That looks right to me. Otherwise, we never extract the variable list
arguments passed in to the ast_variable list update_fields. Yikes.
> I have changed it and it works for me - but -
>
> 1)I don't know what else this may effect
> 2)I dont know how to pass this on to the de
sing):
} else if (!strcasecmp(v->name, "directmedia") ||
!strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
...
Note that these settings and their behaviour is the
gt;
> For subroutines there is the Return() application, but this can not be used
> generally 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 t
+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
ice
represented by extension 1000 in your scenario is handled by Asterisk,
and causes the channel on the other side of the bridge with that SIP
channel to be put on Hold.
There is no mechanism in Asterisk today to pass through a re-INVITE to
initiate a remote hold.
--
M
ugh Asterisk. Clearly,
something is abusing it.
I think 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 di
you want to tell dialplan execution to wait for more digits, you
can use the Incomplete dialplan application [2]. 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_Incomp
27; extension. If you disable 'endbeforehexten',
then you will get a CDR for the channel while it updates the hangup
logic - 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 ent
gt; Defaddr->IP : (null)
> Prim.Transp. : UDP
> Allowed.Trsp : UDP
> Reg. exten :
> Def. Username: 719111
> SIP Options : (none)
> Codecs : (ulaw)
> Auto-Framing : No
> Status : UNKNOWN
> Useragent: Linksys/SPA2102-5.2.10
> R
ould always be
'xxx@vm_context' where appropriate. I don't think this is your problem
however.
> Or maybe some 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?
W
something else monitoring 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://di
ex.com/voffice/onstage/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
85084.html
[3] 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
--
_
. If
you need more assurance that your issue 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.
media (assuming OP meant 'no sound').
I could have missed it in the pastebin, but I didn't see a
request/response from Asterisk that was either sent to a private IP
address or contained a private IP address in the SDP. In the trace
that you provided, which request/respons
y+Code+of+Conduct
[2] 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 &
contacts
> 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
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://as
roduction to ARI 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 - Huntsvil
have DONT_OPTIMIZE 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 - Huntsvil
sterisk.org/wiki/display/AST/Patch+Contribution+Process
[2]
https://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://a
es do
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.
--
Matth
bug tracker and be sure to mention the documentation too
> since that should be updated as well. Asterisk issue guidelines can be found
> at the following [2].
>
> [1] https://issues.asterisk.org/jira
> [2]
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
n be configured in a myriad of ways. The user 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.
--
Matthe
mal 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 - Huntsv
if you provided some information about the ARI
application you've written. Have you connected a WebSocket? 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 & h
sterisk+13+REST+Data+Models#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
--
_
-- Bandw
iki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL
Action: Setvar
Channel: (your channel name here)
Variable: CHANNEL(musicclass)
Value: (your MoH class here)
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: htt
code the media 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/
e flag to
whatever application caused the bridging 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
--
___
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 358
he the unwashed masses you’re planning to do away with the dial plan?
>
This will be the last time I respond to any of your e-mails on the Asterisk
mailing lists or engage with you in any fashion. Your tone, language, and
rhetoric are all indicative of someone who is not interested in havin
owing set in pjsip.conf (snippet):
> type=endpoint
> disallow=all
> allow=g722
> allow=ulaw
> transport=transport-lan
> send_rpid=no
> send_pai=yes
> direct_media=yes
> tos_audio=46
> tos_video=34
>
> Is there something I'm doing wrong here?
>
> Thanks
On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht wrote:
>
> On Oct 22, 2014, at 11:31 AM, Matthew Jordan wrote:
>
>
> On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht
> wrote:
>
>>
>> On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote:
>>
>> > Paul A
cal idea does not mean we
refuse to discuss it.
This is an open source project. Communication is done in an open,
transparent manner. People should feel like they can bring up interesting,
radical, and yes - even crazy - ideas.
If you don't like that, you don't have to particip
}
>
> }
>
> }
>
> If this line:
>
> memset(&cdr->answer, 0, sizeof(cdr->answer));
>
>
>
> is commented away, should work.
>
>
>
>
>
I don't think you need to change the Asterisk source for this (particularly
since it i
On Fri, Oct 17, 2014 at 11:06 AM, A.Santoro wrote:
> On Wed, 15 Oct 2014 09:14:41 -0500, Matthew Jordan
> wrote:
>
> >On Wed, Oct 15, 2014 at 1:50 AM, A.Santoro wrote:
> >> Hi there,
> >> I have installed Asterisk version 12.6 (on Debian wheezy) and I n
produces this behavior?
> I
> am only 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
is
open source. If you'd like to have 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. | En
>
> Variable: CDR(start)
>
> Value: 2014-10-15 11:35:37
>
>
>
> Action: Getvar
>
> Channel: xxx
>
> Variable: CDR(answer)
>
>
>
> Response: Success
>
> Variable: CDR(answer)
>
> Value:
>
&g
his happen both 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
s two
things:
(1) Executes the 'h' extension in the current context, if available
(2) Executes any hang up handler subroutines that were attached to the channel
For more information on the latter, see the documentation on Hangup Handlers:
https://wiki.asterisk.org/wiki/display/AST/Hangu
does not appear as if you have that
buddy/resource combination, in which case 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
rts?
[1] https://issues.asterisk.org/jira
[2] 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 ou
people use it, other than it comes up from time to
time in the issue tracker (which is about the extent of my visibility
for usage).
>> 3) If DUNDi is not really used in modern set-ups, then what are my
>> alternatives?
>>
>> I really have searched and read and Googled everything
Command_get+variable
[2] https://wiki.asterisk.org/wiki/display/AST/AGICommand_set+variable
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
re using a flexible backend (such as
cdr_custom or cdr_adaptive_odbc), you can add a custom column to your
CDR records - such as 'clid_original' - and use the CDR function to
set that value prior to changing the caller ID:
exten => Set(CDR(clid_original)=${CALLERID(num)})
exten =>
can't reproduce this. We've been running a lot of
tests with a variety of SIP clients over the past week here at SIPit -
both with and without ICE - and I haven't had a single instance of
Asterisk failing to provide any ICE candidates when it is properly
configured to do so.
--
Matthew J
work on whatever channel it was set on. If you are
going to use it on an outbound channel, then you should use a pre-dial
handler to apply it to that channel.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
e hood and things
managed to remain the same was the goal.
chan_pjsip does use a different set of rules for how it offers its
codecs, and should generally follow what it outlined on that wiki
page.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsvill
T/Asterisk+Versions
The success of Asterisk 1.8 is due to the involvement and support of
the Asterisk community. As always, thank you for your support of
Asterisk!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://
* much in new versions...)
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by
.9.1).
>
> https://github.com/fail2ban/fail2ban/pulls
>
> HTH,
> Patrick
>
Why would you not use the SECURITY log format, which have the exact same
format between chan_sip and chan_pjsip, and have a consistent format from
Asterisk 10+?
https://wiki.asterisk.org/wiki/display/A
gt; m=audio 18366 RTP/SAVPF 0 3 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=connection:new
> a=setup:actpass
> a=fingerprint:SHA-256
> CE:EE:D9:28:EA:B0:6E:D0
and my sip.conf:
>
> [general]
> bindport = 5070
> bindaddr = PU.BL.IC.IP
> udpbindaddr = PU.BL.IC.IP
> tcpenable = yes
> limitonpeers = yes
> rtcachefriends = no
> tos_sip=cs3
> tos_audio=ef
> realm = testers.com
> autodomain=yes
> domain=PU.BL.IC.IP
> dom
ns you may have
at members of the Asterisk Development Team (myself included).
More information about the hackathon can be found on the ChallengePost page
or at http://www.asterisk.org/community/astricon-user-conference/hackathon
See everyone in Las Vegas!
Matt
--
Matthew Jordan
Digium, Inc
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
people may have had regarding Asterisk and the
UniMRCP project.
Thanks -
Matt
[1] http://www.unimrcp.org/
[2] http://www.gnu.org/licenses/license-list.html
[3] http://svn.asterisk.org/svn/asterisk/trunk/LICENSE
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsvill
es.asterisk.org/jira/browse/ASTERISK-24234
You may want to try the patch on the issue to see if it resolves your
crash. Alternatively, you could try checking out the 12 branch.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
bug. When things got ported over to hit the cached
snapshots of the channels (as opposed to locking the live channel),
that field got missed.
Please file a bug on issues.asterisk.org.
Thanks!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL
that aren't delivered with Asterisk?
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Bandwidth and Co
ch a log are on the wiki [3]
* Execute the CLI command 'core show fd'. This will dump out all
allocated file descriptors. Attach the output of the command to the
issue as well
[1] http://lists.digium.com/mailman/listinfo/asterisk-biz
[2] https://issues.asterisk.org/jira/browse/ASTERISK
1374 would be the port.
>>
>> /Mikael Fredin
>
> Sure but what I'm looking for is to:
> - type something like "rtp show settings"
> - and read something like : Port range 1-2
That information is not available via a CLI command.
--
Matthew Jo
edia.so) ?
>
Does pkg-config find libpjproject?
$ pkg-config --list-all | grep libpjproject
Asterisk's configure script uses pkg-config - so if that can't find
it, Asterisk can't find it.
--
Matthew Jordan
Digium, Inc. | Engineering Manag
nk some more
technical details about Bleep would be helpful for the Asterisk
developers, so we could see what would be needed for Asterisk to
communicate with Bleep.
Thanks!
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us o
On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik wrote:
> Hi
>
> Is anyone using asterisk on CentOS 7?
>
> If so, is it working fine and as expected?
>
Random data point: the Asterisk project's build agents are still on CentOS 6.
Your mileage may vary.
--
Mat
we're thrilled with that 5 second wait
time. See https://issues.asterisk.org/jira/browse/ASTERISK-23259 for a
bug report noting this behaviour.
> Why are you attempting to request an agent that has a device state
> (Agent:agent_id) of busy anyway? That agent could be on another call
with Oracle
use LENGTH, not LEN.
Your solution, as it is currently, wouldn't be acceptable, as it would
cause far more problems than it would solve. About the only way I
could see solving this would be to make it configurable some place.
Given the relatively few number of people who u
t the documentation fixed.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- Bandwidth and Colocation Pro
is seems quite odd.
Keep in mind that asking for help with deployment issues on asterisk-users
is entirely appropriate, but do remember this is an open source project and
everyone who replies on here is doing so of their own volition. No one is
required to solve your issue for you.
--
Matthew
ones
are preferred:
https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
For those who want the cliff's notes version, it is:
* res_timing_timerfd
* res_timing_kqueue (where available)
* res_timing_dahdi
* res_timing_pthread
In particular, res_timing_pthread should only be used as a l
ons are mutually exclusive because GCC places a
> trampoline on the stack.
>
> The lack of NX-Stacks could be a security defect and could lead to
> governance problems.
>
I'm sorry you don't like nested functions.
The use of RAII_VAR has saved the Asterisk project o
showing XXX for pjlib
>
> Please let me know if any more information is needed
>
>
What is the output of "pkg-config --print-provides libpjproject"? For that
matter, does "pkg-config --list-all" show libpjproject as a package?
--
Matthew Jordan
Digium, Inc. | Eng
yet,
unfortunately, was overlooked.
Ideally, it'd be in the CHANNEL function.
If anyone is curious, the accessor function you want is
ast_channel_callid. It returns the callid ref bumped, so you do have
to make sure you decrement the ref count using ast_callid_unref. You
can print the callid to t
res_pjsip_* modules couldn't be loaded
on a particular instance of Asterisk would be helpful.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: htt
which
is a core supported module in Asterisk 12.
[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com &
;
> Should I open a bug or there is something I am missing?
>
I suspect you have some configuration error or environment problem.
The Asterisk Test Suite - which runs on every commit and nightly -
makes extensive use of custom asterisk.conf files to sandbox instances
of Asterisk that run con
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