Ok, let med rephrase this:
Does anyone actually uses Confbridge and what are your experiences?
How about local channels in combination with Confbridge?
/Johan
Den 2015-09-23 kl. 13:37, skrev Johan Wilfer:
Hi!
I did some tests with Asterisk 11.19.0 and Confbridge. I've wanting to
migrate from
Hi!
I did some tests with Asterisk 11.19.0 and Confbridge. I've wanting to
migrate from Meetme to Confbridge for a long time now.
For two participants in a conference - one actual call and one local
channel that are recording - the cpu sits at 20%. The result was the
same with
Hi!
After upgrading kernel from 2.6.32-openvz-042stab108.8-amd64 to
2.6.32-openvz-042stab111.11-amd64 I fail to load the DAHDI kernel modules.
It's the package dahdi-linux-complete-2.10.2+2.10.2 and it looks like it
compiles and installs just fine (a few error in the text indicates that
Den 2015-09-15 kl. 16:52, skrev asandoval...@gmail.com:
Hello Marek! I’ve been running on an issue with my Asterisk 12
configuration for using WebRTC on a LAN environment for about a month! I
really need some help …
My calls from the browser are done fine. I get ringing, they can be
answered
is mostly using the same Kernel api:s that OpenVZ uses,
but OpenVZ also has some cusom stuff.
If you need Dahdi you will need to give the VE's access to these
devices, there are articles out there that explain how this is done.
Good luck!
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.
With proper monitoring of the resources, you can troubleshoot. I've
found voipmonitor.org to be invaluable in that regard.
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'fullybooted'.
What PHP framework/library are you using -- and why?
I use these, works very well:
https://github.com/marcelog/PAGI
https://github.com/marcelog/PAMI
More modern, uses composer, does get updates.
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that works for you.
Good luck!
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://issues.asterisk.org/jira/browse/ASTERISK-20441
If it is this is a bug in the AEL compiler I think.
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catch however.
Good luck!
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is welcome. Thanks!
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2014-04-10 15:02, Gopalakrishnan N skrev:
Thanks Johan. Are you using this application for any credit card processing?
Nope, mostly ChannelRedirects and injecting sounds in conferences /
controlling via webgui. But the libraries are very good and seemed very
complete to me.
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running asterisk on the metal? Or can I expect
roughly the same performance?
Thanks!
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great. PAGI seems to be a sister-project for AGI.
https://github.com/marcelog/PAMI
https://github.com/marcelog/PAGI
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was more
about if I could expect roughly the same performance, or if it is
drastically different with virtual machines on VMware.
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New
was more
about if I could expect roughly the same performance, or if it is
drastically different with virtual machines on VMware.
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New
2014-04-04 21:35, Kevin Larsen skrev:
From: Johan Wilfer li...@jttech.se
Sounds very good. Do you have this experience with WMware in particular
or with virtualization in general?
We run our Asterisk 11 instance in VMWare as well. They share the
hardware with multiple other boxes. We do
- setting quiet=yes still plays join/leave sound. My current
work-around is:
sound_join=silence/1
sound_leave=silence/1
But this seems a bit ineffective... In Meetme the quiet-flag also
disabled join/leave sounds. Is this by design or an oversight?
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2014-04-04 22:01, Johan Wilfer skrev:
Hi,
I'm doing an evaluation of Confbridge (migrating from Meetme). Looking
at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10
Under the heading User Profile Configuration Options the option
announce_only_user is present. The sample config looks
2014-04-04 23:33, Johan Wilfer skrev:
Also - setting quiet=yes still plays join/leave sound. My current
work-around is:
sound_join=silence/1
sound_leave=silence/1
But this seems a bit ineffective... In Meetme the quiet-flag also
disabled join/leave sounds. Is this by design or an oversight
and odds are that it is
already resolved in a later version.
http://www.asterisk.org/downloads/asterisk/all-asterisk-versions
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New
. On some distros irq's are not
balanced by default and are all hitting the same core.
On Debian I had to install the irqbalance package and the load was
spread across the cores.
Dahdi is still a single thread thought.
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for theese calls. I would do packetdumps with
tcpdump and then analyze it with wireshark. I use voipmonitor to do this
(it gives you a pcap for each call), but tcpdump works fine also.
This could be a congested link, a broken media gateway, or anything.
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2013-11-13 11:55, Jonas Kellens skrev:
On 11/13/2013 11:48 AM, Johan Wilfer wrote:
2013-11-12 17:42, Jonas Kellens skrev:
X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%)
0. 000136 00 ( 0.00%) 0.0031
X.X.X.70 68289fc05ff 00:02:27 007318 060143
2013-10-17 17:48, Russ Meyerriecks skrev:
On Thu, Oct 17, 2013 at 7:53 AM, Johan Wilfer li...@jttech.se
mailto:li...@jttech.se wrote:
1. In asterisk can I get the channel-number of the call so I can
have different logic for the different channels?
Sure, I guess I would just create
2013-10-03 09:52, Johan Wilfer skrev:
2013-10-02 17:12, Shaun Ruffell skrev:
On Wed, Oct 02, 2013 at 01:17:15PM +0200, Johan Wilfer wrote:
If I did use the core timers in dahdi (not loading dahdi_dummy) I
got bad quality in the conferences and dahdi_test showed 99.6% as
worst.
Hmm
do I handle answer / hangup with CAS? Will DAHDI keep this
channels up, or should I query the state of the channels (how?) and
bring them up myself if they are down (Dial?)
3. Other suggestions?
CAS is unknown territory for me so I appreciate all the pointers you
have. Thank you!
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.
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2013-10-02 17:12, Shaun Ruffell skrev:
On Wed, Oct 02, 2013 at 01:17:15PM +0200, Johan Wilfer wrote:
If I did use the core timers in dahdi (not loading dahdi_dummy) I
got bad quality in the conferences and dahdi_test showed 99.6% as
worst.
Hmm...this is the first report I've heard
Hi,
I have some servers that are dedicated to do meetme conferencing. From
some previous test i concluded that I need to use dahdi_dummy as it is
more accurate.
If I did use the core timers in dahdi (not loading dahdi_dummy) I got
bad quality in the conferences and dahdi_test showed 99.6%
2013-10-02 13:55, Gareth Blades skrev:
On 02/10/13 12:17, Johan Wilfer wrote:
Hi,
I have some servers that are dedicated to do meetme conferencing. From
some previous test i concluded that I need to use dahdi_dummy as it is
more accurate.
If I did use the core timers in dahdi (not loading
http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.html
Google :-)
/J
2013-09-27 17:47, Daniel van den Berg skrev:
Hi Asghar,
How do I search the site as I dont see a search bar anywhere...could you
please give me the link to the solution in the list or educate me on how
to
at asterisk's func_odbc and maybe also database replication.
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-port PCI-E E1 card, wct4xxp
driver. The symptom was choppy sound for all calls. The problem was
resolved by disabling hyper-threading.
From that moment on I always disable HT, to avoid problems.
I havn't seen your specific 12.5% cpu problem thought.
Maybe worth a shoot anyway?
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On Fri, Jul 19, 2013 at 2:52 PM, Johan Wilfer li...@jttech.se wrote:
2013-07-19 15:35, Thiago Coutinho skrev:
Hi all.
I'm trying to limit the number of participants in a conference room
with the realtime option maxusers, but it doesn't work.
Someone have this option working properly?
Try
://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT
This is how I do it. This way you can do it more flexible in the dialplan.
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2013-07-01 15:04, Daniel-Constantin Mierla skrev:
Hello,
On 6/28/13 4:29 PM, Johan Wilfer wrote:
Hi,
We have some Asterisk servers that we are moving behind a NAT to
preserve public addresses and make room for growth. This is Asterisk 1.4
NAT works very good with the externip/localnet
-solid and no crashes at all, but no transfers either.
Maybe you can use 32-bit with PAE? This way you can use up to 64G memory
in the server, but maximum 4G per process.
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, but any input
on the topic is welcome. If this is supported in later versions we can
maybe work around until we migrate later.
Thanks!
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+
confbridge? This way you won't need DAHDI.
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http
would not
even compile Meetme if DAHDI header files isn't present.
So you must have installed DAHDI at some point?
Try run dahdi_test (after installing), you should get output like 99.9%
99.8% ... and so on. Then you know the timing works (dahdi psuedo)
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should
do development and testing on a dedicated server. After that, try move
to a VPS and try to fix the issues. The other way around is very hard.
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http://stackoverflow.com/questions/11592010/compile-dahdi-on-openvz-vps-kernel-issue
Good luck!
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2013-05-02 13:19, Marie Fischer skrev:
Hello everybody,
from time to time, we get so-called simplex / one-way audio calls, where one
party cannot hear the other. The only thing in common is that is does happen
with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in
Please, excuse me but I'm not sure I got your suggestion and I'm
realizing I didn't correctly describe my lab set up.
At the moment, the router between both servers provides Internet access
to server1.
That means it has one WAN interface eth0 which is on server2 side and
one eth1 LAN
This means all traffic to 4.3.2.1 will go to dev eth0 on your router.
(The device in your network with the ip 4.3.2.1 also needs to have a
route back to your router for replies.)
Good luck!
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-January/277342.html
http://media.ccc.de/browse/congress/2010/27c3-4193-en-having_fun_with_rtp.html
So my guess is that if you get two devices using the same port, or one
device that don't stop sending, you will hear that injected in your call.
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, and they
look very promising. I've used the TE420/TE820-cards previously, and
they just keep working.. :-)
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will use the proxy:
SIP: Provider - (vpn/tls) - Kamailio - (udp) - Asterisk
RTP: Provider - -- - Asterisk
Good luck!
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, and they are asking for our
recommendations on gateways.
Your advice on this topic is very appreciated!
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-list of doom.. :-)
Maybe you should check it out?
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that are highly appreciated..
You can do all this with Meetme, ChanSpy and ChannelRedirect. You could
also use the AMI variants of the above commands. Maybe you could use
Confbridge that is intended to replace Meetme.
So the simple answer is yes, this can be done.
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, but I think that will get you started.
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(or in the safe_asterisk-script or the
init.d-script).
I think this is per default 1024 on debian, and if you use sip + meetme
you will hit the limit with about 150 concurrent calls.
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JT Technologies Telecommunications AB
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2012-11-08 00:26, Jeff LaCoursiere skrev:
On 11/07/2012 05:20 PM, Jeff LaCoursiere wrote:
On 11/07/2012 02:16 PM, Johan Wilfer wrote:
2012-11-07 20:49, Jeff LaCoursiere skrev:
Just to chime in, if you REALLY want multi-tenant, it is super easy and
surprisingly efficient to use kernel level
)?
Distribution?
Any other pitfalls or recommendations with LXC?
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) than the build in timing in dahdi that is used if
you don't load dahdi_dummy.
Good luck!
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tried goto? I have some extensions that are related and I use
goto to the main context from the others.
Goto(context,h,1)
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2012-09-07 16:13, David M. Lee skrev:
On Sep 7, 2012, at 1:49 AM, Johan Wilfer wrote:
Hi!
I'm trying to limit the permissions for a AMI-account. But I'm a little bit confused by
the permissions. The commands I use are (output from manager show commands,
btw: privilege col seems cropped
!
Tony
Maybe a I'm reaching here but.. I had some very strange issues with
broken quality with Monitor and a NFS mount. This was 1.4, but several
years ago. I ended up not using NFS in the end.
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is:
username: pbx_ami
secret: Set
acl: yes
read perm: none
write perm:
system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,all
displayconnects: yes
Can someone explain this please?
Thanks!
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/${dir}
for i in ${src}/*.wav; do sox $i -V -r 8000 -c 1 -q -s \
${dst}/$(basename $i .wav).wav vol 0.8; done
normalize-audio -a -20dBFS ${dst}/*
done
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2012-08-28 16:44, Andrew Latham skrev:
On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote:
Hi,
I've used the shells-script at the end of this email to generate 8khz mono
wave-files for asterisk from a 144 khz recording.
Try this to test with
http://www.digium.com/en
2012-08-28 17:04, Andrew Latham skrev:
Yep, check out repotools for that
http://svn.asterisk.org/svn/repotools/sound_tools/scripts/
Cool! Thank you!
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2012-08-27 19:48, Markus skrev:
Hi Matthew,
Am 27.08.2012 15:41, schrieb Matthew Jordan:
When they adjust the volume of the stream, if effects only their
stream,
and not the volume of the stream of the other callers.
In short: All callers at all times are *always* in the same
conference,
but
(chan_alsa again) to
monitor the connection and if the channel is not there to
hangup?
In sip.conf you could use rtp-timers to hangup a call if the
media-stream stops to flow.
Look at these options in sip.conf:
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=0
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2012-04-09 22:32, Johan Wilfer skrev:
2012-04-09 20:22, Carlos Alvarez skrev:
On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
At first, if your Asterisk is in a VM install it on the real
server, it solved us on some installations
can eliminate this manual and
time-consuming process in the future. And know about the problems before
the customer complains about the quality..
Thanks in advance!
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, and
configured the switch to duplicate the traffic to the quality server? Or
do you run this on the same server as asterisk?
Thanks for the suggestions!
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in the
en_baselevel-directory. After that it will look in the en-dir.
Can't find the docs for this right now but this way you don't need to
copy all the recordings, and you can stack as many layers as you like.. :-)
/Johan
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usually ignore this error.
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if
there are any issues or special methods required to get dahdi running
(such as the DEVNODES feature in vz).
Also, sorry to anyone if I've veered too far offtopic, I'm quite
interested and invested in openvz/asterisk/dahdi interoperability.
I'll keep you updated! Thanks for you input!
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. The
architecture is AMD64-compatible and Debian AMD64 will run on AMD and
Intel processors with 64-bit support. Because of the technology
paternity, Debian uses the name AMD64.
I've always wondered about the amd in the name, but it makes sense
now. Thanks for the input..
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% 99.996% 99.996%
99.997% 99.992% 99.994% 99.986% 99.998% 99.991% 99.987% 99.991%
99.999% ^C
--- Results after 201 passes ---
Best: 100.000 -- Worst: 99.962 -- Average: 99.994760, Difference: 99.998428
Thanks!
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JT Tech | Developer
should use another
distro for the HN.
Maybe I will try switch to lxc instead of openvz as it is in the
mainline kernel now. After all I need two things: Isolation, and the
possibility to run multiple asterisk VEs on the same physical machine.
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amd64 be responsible for the high cpu in kernel mode?
- I have a spare Digium TE220, would it offload the server to use it as
a timing source only?
- How do I debug the high cpu usage by the kernel, can I break this
down by module in some way?
Many, many thanks!
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to rebuild everything on i386
architecture, but that's the last resort.
/Johan
On 1/18/2012 4:24 AM, Johan Wilfer wrote:
I'm in the process of replacing an old server with a new one and are
making som changes in the infrastructure, the biggest change in my eyes
is moving from i386 to AMD64 arch
On 1/18/2012 8:52 AM, Johan Wilfer wrote:
2012-01-18 11:31, John Knight skrev:
Hi Johan,
I've run into a similar issue before. I didn't resolve the problem
per se, but I got around it by modifying modules.conf to disable the
loading of res_timing_timerfd.so and loaded res_timing_dahdi.so
2012-01-18 17:50, Shaun Ruffell skrev:
On Wed, Jan 18, 2012 at 02:52:47PM +0100, Johan Wilfer wrote:
2012-01-18 11:31, John Knight skrev:
Hi Johan,
I've run into a similar issue before. I didn't resolve the problem
per se, but I got around it by modifying modules.conf to disable
2012-01-18 20:06, Shaun Ruffell skrev:
On Wed, Jan 18, 2012 at 12:58:31PM -0600, Kevin P. Fleming wrote:
On 01/18/2012 12:15 PM, Johan Wilfer wrote:
2012-01-18 17:50, Shaun Ruffell skrev:
One question first though, is your new server able to keep accurate
time with nt, or is the clock
and install
asterisk.
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configuration.
I'm in the process of building another server with openvz, so I'll need
to refresh my memory and try to document the procedure.
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the 2nd instance to
listen on another port?
I've used openvz for this, it's not realy virtualisation - all the
virtual machines (asterisk-boxes) share the same linux kernel.
If you want to use dahdi/meetme you will have to let the VE's use the
/dev/dahdi devices but thats not very hard.
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you checked to documentation in the CLI? The J option is something
different..
/Johan
On Fri, Jun 10, 2011 at 12:23 AM, Johan Wilfer li...@jttech.se
mailto:li...@jttech.se wrote:
Humm... Seems like my message didn't make it. Here we go again..
/Johan
Original Message
Humm... Seems like my message didn't make it. Here we go again..
/Johan
Original Message
Subject:Re: [asterisk-users] ControlPlayback's options
Date: Sun, 05 Jun 2011 22:19:18 +0200
From: Johan Wilfer li...@jttech.se
To: Asterisk Users Mailing List - Non-Commercial
On 2011-06-04 13:38, virendra bhati wrote:
Hi Johan Wilfer,
Thanks for your reply. On the basis of your provided code I made all
things into extensions.conf. But i have an small issue on which I need
your attention again.
in below context what's ${tz} ? Is this time zone value or else?
Yes
this in your dialplan...
I've attached some ael I use for this to implement 1 and 3 as 1 minute
rewind/forward. 4 and 6 as 5 minutes rewind/forward and 7 and 9 as 15
minutes.
5 I use as the pause key, and */# to switch recording.
Greetings,
Johan Wilfer
context
handling that was fixed between Debian 5 and 6.
Maybe you experience something similar?
/Johan
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