, chan_ooh323 from my side to
another asterisk SIP chan_sip on both sides.
Just because everything work OK, I , definitely, can comment out this error
message, but...
Could you give me any idea why this error can appear?
If you haven't create an issue on Jira, this is a bug.
--
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here. We also do this and it works quiet well. Kudos.
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It should be noted, we did have a FreeBSD and Ubuntu systems running
the testsuite back in 2010. FreeBSD was donated to the project.
I personally had a PowerPC system running asterisk / testsuite, on debian.
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storage device.
Alternatively, you could use something like Puppet to deploy the files to
all the servers.
This is basically what we do, we use puppet to help distribute files
to remote servers while still using app_queue. Shared network drive
also works.
--
Paul Belanger | PolyBeacon, Inc.
Jabber
phase.
You should be able to google Asterisk dialers to see some example that
people have done.
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On Wed, Jan 28, 2015 at 1:37 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for
queues.
For a particular customer, when I run queue show
as to what it means?
Thanks in advance
Welcome to business logic embedded into app_queue. The issue with the
queue show command rendering stats, is what timeframe are the stats
aggregated over? IIRC, the calculations are using a moving
average[1].
--
Paul Belanger | PolyBeacon, Inc.
Jabber
chrome.
I hope someone can intersperse the output with comments?
Pastebin the fill debug, you've delete an important piece of information.
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cause = ) in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6]
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
Can anyone tell me how this should be used ?
sip.conf: storesipcause=yes
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an external process?
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_
-- Bandwidth and Colocation
with the other user?
Since what you describe is a valid for SIP, you'll have to drop the
packets at the network level (firewall). Or use the ACL system in
asterisk to restrict it.
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to check if there's any chance I could ask Asterisk not to
register when I reset. Or is there any other possible solution for this?
No, only reload after your ITSP brute force timer has expired.
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. I also believe there are some open issue with dtls +
srtp too.
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, in your setup listed
above, rtpengine is not needed, since newer versions of asterisk
support both. Adding it in will just complicate your setup.
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information[1]
on the web, you just need to google for it.
[1] http://www.slideshare.net/crocodilertc/webrtc-websockets
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how to handle this scenario in the new versions as
well, I'll probably need to upgrade ahead anyway.
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or something else ?
libpjsua.so (libc6) = /usr/lib/libpjsua.so
You will likely need to pass the pjproject directory to configure.
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a
kamailio peer and away you go.
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--
_
-- Bandwidth
seconds.
It's only happened once in 2 years that I know of, so may not be worth
worrying about.
AMI will raise the AgentCalled[1] event.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_AgentCalled
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On Thu, Apr 17, 2014 at 7:25 AM, binary dreamer
dreamer.bin...@gmail.com wrote:
hi. I would not do that due to network issues.
My approach is to record everything locally and every hour or so to move
everything to a storage.
+1 save yourself the headache and do this.
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Paul Belanger
could leverage snapshots in VM ware for the purpose or migrating or
back ups. I don't think it is a waste per say, just different
requirements.
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.
Or can you express your creativity by fiddling with ASTERISK_PROMPT?
If you really want to do it:
1) create a wrapper to asterisk -r
2) pipe the welcome message to /dev/null
3) ???
4) profit
you didn't modify Asterisk.
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the site now. Currently
only the 3rd edition is published online.
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want to use. We used starpy for a
while, but ended up rewriting our own version. Currently we're
connecting AMI to a message bus and passing events across the bus.
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, now when a user changes their password, secret.conf gets
updated not voicemail.conf.
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: AppVoicemail
;! Creation Date: Thu Mar 20 06:48:16 2014
;!
i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not
using realtime.
anyway to prevent AppVoicemail ro auto generate files?
passwordlocation = spooldir
Read voicemail.conf about how to use it.
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to change it with modprode.
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--
_
-- Bandwidth
are you looking for? We did some load
testing recently and found less people in a bridge is better then
more. Audio source location didn't really matter much.
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goes live.
Correct, in this case para-virt is not the way to go. You'll want to
use a virtualization platform that does support multi-hardware with
live migration support.
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,
auto-scaling VoIP setup. :)
+1 to this post. A lot of good information here.
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On Thu, Feb 27, 2014 at 10:55 PM, Darryl Moore dar...@moores.ca wrote:
On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:
No such thing as 'free open source g729 license', if you actually read the
site:
There is regarding the copyright on the code. The fact
material on the subject however,
I am still in need of some definitive answers.
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,hangup,1)
-- Executing [hangup@app-blackhole:1]
NoOp(SIP/trunk503in-010b, Blackhole Dest: Hangup) in new stack
-- Executing [hangup@app-blackhole:2]
Hangup(SIP/trunk503in-010b, ) in new stack
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. If you offer a both g729 and ulaw, then ulaw will be used.
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of a Linux support issue then specific to
Asterisk. Depending on your OS, will dictate how to change your
gateway.
check /etc/network/inferfaces if you are ubuntu / debian.
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On Thu, Feb 13, 2014 at 1:04 AM, George Joseph
george.jos...@fairview5.com wrote:
On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote:
Hello,
How does extensions.lua compares
or memcached.
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--
_
-- Bandwidth and Colocation
away from your core Asterisk box. I
suggest picking up the book[1] and reading the chapter on connecting
multiple Asterisk boxes together.
[1] http://www.asteriskdocs.org/
--
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On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
and arriving at the Asterisk server. This is why it's a mystery that
Asterisk doesn't see the call coming in. We tried
On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
and arriving at the Asterisk server
, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
At this point in time, you'll need to show us a .pcap on the Asterisk
box, when you make a call to it via Kamailio.
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Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan
to be reliable. If you need fax, you should
be using T.38. Your codec is likely the issue.
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.
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--
_
-- Bandwidth and Colocation Provided
hop across your network. If are
Asterisk is not getting anything, either it is not receiving anything
(check transmit side) or the firewall is dropping it.
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and on).
TEST_FRAMEWORK is an option selectable under the Compiler Flags -
Development menu in menuselect.
./configure --enable-dev-mode
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and insist using SIP. If your ITSP cannot
accommodate your request, thank them and look for another provider.
H323 is Asterisk is basically dead, sure there is a module, sure it
might compile, but you'll be going down the path of zero help.
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Jabber
logs and
see what Asterisk is doing when the odbc connection is down. EG: it
should be attempting to reconnect.
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, if you want anybody to call you, you need to leave it open to
the public. Meaning, you can't really secure it. Obviously, don't
have any outbound trunks configured on the box so that the only
location some could dial would be your extension.
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Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan
, clocking, etc.
In the end I had to upgrade dahdi to 2.7+ and the issue went away.
Never did figure out the real problem, but to this day I think the
issue was a delay on the frames from the PCI bus into the software.
All that to say, try upgrading DAHDI and see what happens.
--
Paul Belanger
max,
and asterisk
disconnects the call. We have no idea why this is happening. SIP and RTP is
flowing as
expected.
Your help is greatly appreciated,
Nick.
Show us the problem, give us a SIP trace[1].
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Paul
management/control protocols to
this
list: ARI, and the ExternalIVR interface.
If not, it might be instructive to learn why!
Would also like to see this update to include ARI. We talked a little
about it at astridevcon, and I think it is likely an oversight.
--
Paul Belanger | PolyBeacon, Inc
?
Options 1 - log the agent out, they don't get the next call.
Option 2 - Set up weights for your agents, as answer a new call,
increment then up so they don't get the next.
Either way, I see issues with the setup. Best ways is to rethink your
queue strategy and stop using ring all.
--
Paul
footing on performance. I don´t mean another
slow cygwin port, I man a native Asterisk for windows. In fact, I
would invest on the project if somebody wants to do it.
Do you just sit around and think shit up to blame Digium all day?
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Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan
entered during prompt)
exten = _X,2,Goto(project,s,1)
Then you have a DTMF issue, Background will allow DTMF to interrupt the
prompts.
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tps_processing_function started at [ 468]
taskprocessor.c ast_taskprocessor_get()
55 threads listed.
First thing, prune your Asterisk configuration and don't load any
modules you don't need to use. Are you really using chan_mgcp,
chan_skinny, res_calender, etc.
--
Paul Belanger | PolyBeacon
,s,1)
my problem when the customor call the number 600 and press 1 in order to go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan
there is some trancoding when using voicemail...
How can I find out if there is trancoding ??
Maybe explain what your dialplan is doing. Are you making system calls
to a database or AGI?
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Github
. Then make an educated guess about what is
happening.
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On 13-11-13 10:20 AM, Jonas Kellens wrote:
Hello,
can I use include-statements in the calendar.conf configuration file ?
You _should_ be able to use it will every .conf file, otherwise it is a bug.
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.
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--
_
-- Bandwidth and Colocation Provided
the issue at the source. Spend the money for a UPS at
each desktop, convert your phones to PoE and install a UPS in your
server room.
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https
for me (no affiliation). Or
maybe your Internet link sucks and you need to change your ISP.
^ this
Like others said, you really need to drill down and find out where your
audio issues are. Local is easy to do, since you control the network,
remote is harder.
--
Paul Belanger | PolyBeacon, Inc
is was *[CallerIDnum]
*
*So 'n' is now 'N'
*
Asterisk AMI got basically a rewrite[1] of how it works, so there are
some breaking changes moving forward.
Read ChangeLog and UPGRADE.txt in the source tree for more information.
[1] https://wiki.asterisk.org/wiki/display/AST/AMI+1.4+Specification
--
Paul
On 13-10-21 10:39 PM, bilal ghayyad wrote:
Hello;
I am looking for calls recording solution to do recording based on the network
traffic .. The solution to be competitive and appreciate if it is open source
.. Any suggested one?
http://www.orecx.com/
--
Paul Belanger | PolyBeacon, Inc
, you lost the connection. Open the
connection again and profit.
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about it, I'll step up and pay for the patch. No need for
you to waste your profits on something this.
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?
Eliminating translation is difficult. How do you know you were
successful? Do 'module show like codec_' and 'module show like format_'
show anything unexpected?
Also drop Apache and Database from your PBX.
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the
thread in Asterisk, which is causing your autoservice errors (and yes,
they are real errors) which increases the CPU on asterisk.
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something like iotop, netstat and see what your system is doing.
I doubt this is a CPU issue.
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metrics.
[1] http://www.sevana.fi/aqua.php
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On 13-05-10 02:45 PM, James Mortensen wrote:
I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS
codec, which is part of the WebRTC standard as the default codec.
Doubt it.
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/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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value)
before addressing hardware resources?
Your help is greatly appreciated,
Nick.
You failed to say what happens when 92 channels are created. Show us
your errors.
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Github: https
SIP/2.0
I studied the source code and found no ways to implement it :(
Dmitriy.
How about:
exten =22,n,Dial(SIP/skype.ippi.com!lo...@skype.ippi.com,60,rS(1200))
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with
support asterisk as tts server.
Amit--
Asterisk is not a TTS server, it is a PBX. I'm sure you could hack
stuff together, but you'd be better off to use external services for TTS.
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Paul Belanger | PolyBeacon, Inc.
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/Events from 50 agents?
You don't want to use PHP for your daemon, change to another scripting
language (EG: python).
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of
the existing toolsets. Sadly, documentation is weak, and I don't
suspect it gets much love in production.
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Anybody using Apache to proxy HTTP traffic to Asterisk HTTP? I got a
request from a developer to add some CORS headers[1], for an
application we are writing, and wanted to see if anybody else has had
success.
[1] http://enable-cors.org/
--
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Jabber: paul.belan
-cdr-rollover.
Why not use logroate?
$ man logrotate
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On 12-12-04 10:02 AM, Danny Nicholas wrote:
IIRC log rotate only rolls the files in /var/log/asterisk, not
/var/log/asterisk/cdr-csv
You need to configure logroate with the path and filename.
--
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why you choose to rewrite logrotate :)
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On 12-11-17 06:23 PM, Mitch Claborn wrote:
Is there an Asterisk repository for Ubuntu that has recent versions
(e.g. 11)? The standard Ubuntu repository for Ubuntu 12.04 is stick at
1.8.
None that I know of.
--
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On 12-11-08 01:41 AM, martin f krafft wrote:
also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.07.2340
+0100]:
What is your point of pain? Right now we do most of the
configuration, provisioning, and system management outside of
asterisk.
My systems are already managed
/ puppet). This will help you
get on the right path.
[1] https://github.com/kickstandproject/astricon-2012-presentation
[2] http://goo.gl/T8lJR
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/debian/ast_config
[2]
https://github.com/kickstandproject/astricon-2012-presentation/tree/master/debian
[3]
https://github.com/kickstandproject/puppet-modules/tree/master/modules/asterisk/manifests
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not functioning correctly indicates that a signal
message was unable to be delivered to the remote party; e.g., a physical
layer or data link layer failure at the remote party or user equipment
off-line.
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On 12-09-26 11:12 AM, motty.cruz wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Wednesday, September 26, 2012 7:52 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk
is it that the phone is refused only when writing the override file?
Note that the only logging difference between a successful and unsuccessful
write is the above line from the message log. The tcpdump looks the same.
Permissions issues?
If you switched to FTP or HTTP does it work?
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, I'll share the link when finished
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On 12-08-28 10:14 PM, Chris Nighswonger wrote:
Are there deb packages available for Asterisk 10 or for 11 beta?
None.
Well, the Debian VoIP team has an experimental[1] package for Asterisk 10.
[1]
http://anonscm.debian.org/viewvc/pkg-voip/asterisk/branches/experimental/
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community still see value in the
old mantis data, could we not hand it over for somebody else to manage?
I'm sure we could find somebody to donate the bandwidth.
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On 12-08-22 02:04 PM, Giuseppe Longo wrote:
Just a little questions, what's the difference between asterisk 1.8
and asterisk 11?
Not a little answer[1].
[1] http://svnview.digium.com/svn/asterisk/branches/11/CHANGES?view=markup
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?
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graceful in an
automated way?
Monitor the events on the AMI, you should see the following:
Event: Shutdown
Privilege: system,all
SequenceNumber: 0
File: asterisk.c
Line: 1773
Func: really_quit
Shutdown: Cleanly
Restart: True
Then you can build out your monitoring tools from it.
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Paul Belanger
://svnview.digium.com/svn/asterisk/team/oej/sip-compliance/asterisk-sip.txt?view=markup
Interesting, never knew this existed. I think it would be worth the
time and effort to get this merged into trunk or into the wiki. A great
piece of documentation.
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/debian/ast_config
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mailbox. That way, you can then breakout each mailbox into separate
config files with include statements.
[1] http://svnview.digium.com/svn/asterisk?revision=225406view=revision
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into 1.8.13.2 for Debian?
We don't need to release a 1.8.13.2 release of Asterisk. Once the issue
has been fixed in the 1.8 release branch, it would just be back-ported
into a Debian patch for the package.
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by Tzafrir.
Is anyone officially working on this particular problem already? I was
tempted to have a closer look at it, but don't want to duplicate an
effort that is already underway elsewhere.
Best to check JIRA and see. Actually, does the issue even exist in JIRA?
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Paul Belanger | PolyBeacon, Inc
might want to reach out to #asterisk-dev
or asterisk-dev mailing list.
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