Hello friends.
I have a softswitch in which I cannot create a list of blocked source
numbers; So, I have thought to use Asterisk and return a 302 message
when the number can make the call, my dialplan is as follows:
[from-external]
exten => _AX.,1,Verbose(===> ${CALLERID(num)} to ${EXTEN})
On Wed, 17 Jan 2018 09:26:28 -0700
John Kiniston wrote:
> use func_odbc, create a new function that does a lookup.
>
> [CALLERID]
> prefix=LOOKUP
> dsn=MyDB
> readsql=SELECT CALLERID from MyNames where CallerIdNum =
> '${SQL_ESC(${ARG1})}'
>
> exten => s,n,Set(CALLERID(NAME)=LOOKUP_CALLERID(${C
On Wed, 17 Jan 2018 12:08:40 +0100
Antony Stone wrote:
> On Wednesday 17 January 2018 at 11:59:21, Paul Neuwirth wrote:
>
> > Hello group,
> >
> > I tried a lot to enlarge the frequency (i.e. more announces, low
> > wait between). according to config, every 3
Hello group,
I tried a lot to enlarge the frequency (i.e. more announces, low wait
between). according to config, every 30 seconds the announcement should
take place. In fact, the first periodic announce is done after 2
minutes?
What is my fault?
Thank you
Regards
Paul
# zypper if asterisk
On Tue, 16 Jan 2018 18:18:18 +0200
Tzafrir Cohen wrote:
> On Tue, Jan 16, 2018 at 11:05:01AM +0100, Paul Neuwirth wrote:
> > Hello group,
> >
> > what is the preferred method to connect to asterisk cli over
> > network? I need to run asterisk cli commands remo
Hello group,
what is the preferred method to connect to asterisk cli over network? I
need to run asterisk cli commands remotely.
Sharing the unix socket through NFS, if that's working?
Or any other approaches, despite using SSH or rlogin, rsh.
Thank you
Anyone?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To
s incoming events/response across multiple processes
(multiple AMI connections on the same asterisk machine), should the
ami events/response should be pushed into RabbitMQ so the proess can read
from RabbitMQ ?
Thanks
Paul
--
_
-- Ban
On 15/06/15 07:46, lu...@sulweb.org wrote:
Hello all,
Given the requirements above, what's a cheap but working PCIe card / USB
adapter I could buy for this kind of PBX? Do I need things like echo
cancellation? Do I need FXS ports?
Thanks in advance,
Lucio.
I would get hold of some lower-pow
x27;: Invalid UTF-8 string.
> [Mar 5 11:13:29] ERROR[3526]: json.c:704 ast_json_vpack: Error building
> JSON from '{s: s, s: s}': Invalid UTF-8 string.
>
>
> This is call from H323, as I know avaya , chan_ooh323 from my side to
> another asterisk SIP chan_sip on both sides
an Asterisk 1.8 server
> for webrtc capabilities (but not any other sip). It uses the dispatcher
> module to dispatch to the underlying asterisk so you will still need to add
> the Asterisk to the dispatcher config.
>
+1 to everything here. We also do this and it works q
>
> [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
>
It shou
://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> Hi
>
> All of that is possible and is exactly what we do, both for customer sounds
> and for call recordings. Just make sure you have resilience in your shared
> storage device.
>
> Alternatively, you could use somet
On Wed, Jan 28, 2015 at 1:37 PM, Paul Belanger
wrote:
> On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik 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 ru
the queue show figure wrong due to a bug or am I making an incorrect
> assumption 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
aggreg
looking,
likely, but you should be able to see anything you missed in your
testing phase.
You should be able to google Asterisk dialers to see some example that
people have done.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Githu
ut does show RTP flows to chrome, but there's no sound
> from chrome.
>
> I hope someone can intersperse the output with comments?
>
Pastebin the fill debug, you've delete an important piece of information.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeac
On Oct 29, 2014, at 4:26 PM, Matthew Jordan wrote:
> On Wed, Oct 29, 2014 at 2:31 PM, Paul Albrecht wrote:
>>
>> On Oct 28, 2014, at 5:03 PM, Ben Langfeld wrote:
>>
>> On 28 October 2014 19:47, Derek Andrew wrote:
>>>
>>> What is the alternativ
On Oct 29, 2014, at 2:45 PM, Ben Klang wrote:
>
>> On 10/28/2014 06:03 PM, Ben Langfeld wrote:
>>> On 28 October 2014 19:47, Derek Andrew wrote:
>>> What is the alternative to the dial plan? Is everyone talking about getting
>>> rid of the statements like:
>>> exten => s,1,
>>>
>>> what is t
ecuting [h@pbx-routing:5]
> NoOp("SIP/SipAT01-0015", "sip 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
built for us to move completely away from AMI/AGI.” or this "Paul: take
away apps, and whatever is in the core is what we should care about.”
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://
g anything like:
# asterisk -rx 'core show channels'
via an external process?
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
__
share their vision with the rest of
the Asterisk community.
On Oct 27, 2014, at 2:32 PM, Jeffrey Ollie wrote:
> On Mon, Oct 27, 2014 at 2:04 PM, Paul Albrecht wrote:
>>
>> The reason the dial plan can never be deprecated is because Asterisk
>> wouldn’t be Asterisk without the
dude.
On Oct 24, 2014, at 11:48 AM, Jeffrey Ollie wrote:
> On Fri, Oct 24, 2014 at 10:09 AM, Paul Albrecht wrote:
>>
>> When Matt says deprecating the dial plan would be difficult and would take a
>> long time it seems to me he’s being evasive and misleading. He doesn’t
I have released an updated AppKonference that compiles with Asterisk 13. You
can download the latest code from source forge:
sourceforge.net/projects/appkonference
That said Asterisk 13 doesn’t get that much attention because I use Asterisk
1.4 + some hacks. Here’s a link to my Asterisk 1.4 gi
On Oct 23, 2014, at 1:58 PM, Kevin Larsen
wrote:
> > From: Paul Albrecht
>
> > Seems like now is as good a time as any to raise these issues, in
> > fact, sooner is better than later because once developers start down
> > a path it’s very difficult to get them c
On Oct 23, 2014, at 1:55 AM, Olle E Johansson wrote:
> It is critical that a group of developers ask themself questions along
> these lines - "what if???"
>
> - What if we removed AGi and AMI?
> - What if we made a pluggable PBX?
> - What if we restarted working on a SIP channel?
> - What if w
On Oct 22, 2014, at 3:39 PM, Matthew Jordan wrote:
>
> 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
On Oct 22, 2014, at 3:27 PM, Kevin Larsen
wrote:
> > From: Paul Albrecht
> > Here’s a link to the minutes: https://wiki.asterisk.org/wiki/
> > display/AST/AstriDevCon+2014
> >
> > It has you saying: Leif: we're in a transition, moving from dialplan
s to be some more justification for such a profound change to a mature
product interface than some vague desire by unknown persons who know best for
the entire Asterisk community.
> So, to answer your question, yes, and no.
>
> On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht wrote
On Oct 22, 2014, at 2:26 PM, Leif Madsen wrote:
>
>
> On 22 October 2014 14:55, Paul Albrecht wrote:
>
> On Oct 22, 2014, at 11:31 AM, Matthew Jordan wrote:
>> This is an open source project. Communication is done in an open,
>> transparent manner. People shou
On Oct 22, 2014, at 11:47 AM, BJ Weschke wrote:
> On 10/22/14, 12:14 PM, Paul Albrecht wrote:
>> On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote:
>>
>>> Paul Albrecht wrote:
>>>> Really? Shouldn’t something this major affecting the entire Asterisk
>&
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 Albrecht wrote:
> >> Really? Shouldn’t something this major affecting the enti
On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote:
> Paul Albrecht wrote:
>> Really? Shouldn’t something this major affecting the entire Asterisk
>> community get discussed on the lists? Any idea what Leif is talking
>> about when he says the community is in transition, mov
Really? Shouldn’t something this major affecting the entire Asterisk community
get discussed on the lists? Any idea what Leif is talking about when he says
the community is in transition, moving from dial plan model to external control?
Here’s a link to the notes posted on the Asterisk wiki:
h
t cannot use same username if that username is connected
> 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.
--
Paul Belanger | PolyBeacon, Inc
Mitch,
Is it the below error?
if ((fd = open(filename, O_RDONLY)) < 0) {
ast_log(LOG_WARNING, "Cannot open file '%s' for reading: %s\n",
filename, strerror(errno));
return NULL;
}
Regards,
Paul
__
Hi,
I am mostly concerned with inbound calls.
Would it work the same?
Regards,
Paul
From: asterisk-users-boun...@lists.digium.com
on behalf of Gopalakrishnan N
Sent: Monday, August 18, 2014 4:13 AM
To: Asterisk Users Mailing List - Non-Commercial
...
Any advice would be most welcome!
Regards,
Paul
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.as
t think that
> I'm brute forcing.
>
> Just a question 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.
--
Paul Belanger |
On Fri, Aug 15, 2014 at 12:17 PM, Olli Heiskanen
wrote:
> Thanks Paul, I appreciate your thoughts.
>
> I understand your way, it's logical in your environment. I prefer to use LTS
> versions of Asterisk so I'm guessing what I want to do is not quite possible
> with Aster
And, for us, we
keep RTP/SAVPF outside of asterisk since support for it has been
recently added. I also believe there are some open issue with dtls +
srtp too.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: p
Thursday, August 14, 2014 03:15:16 AM Paul Greenberg wrote:
> Hi Anthony,
>
> That script does not work. My guess is that it is related to the way
> asterisk interacts with CentOS environment.
>
> Best Regards,
> Paul Greenberg, Esq.
>
> On Wednesday, August 13, 2014 12:1
Hi Anthony,
That script does not work. My guess is that it is related to the way asterisk
interacts with CentOS environment.
Best Regards,
Paul Greenberg, Esq.
Law Office of Paul Greenberg
530 Main Street, Suite 102
Fort Lee, NJ 07024
Tel: 201-402-6777
Fax: 201-301-8876
Web: http
On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen
wrote:
> Hi,
>
> Wow, thanks Paul, realizing the problem makes a lot of sense.
>
> So I setup Kamailio as a peer, but if I disable chan_sip module completely,
> I can't do it in sip.conf like I'd otherwise assume to do. I
manually.
Best Regards,
Paul Greenberg, Esq.
Law Office of Paul Greenberg
530 Main Street, Suite 102
Fort Lee, NJ 07024
E-mail: p...@greenberg.pro
Tel: 201-402-6777
Fax: 201-301-8876
Web: http://www.greenberg.pro
From: asterisk-users-boun...@lists.digium.com
r words should I get the line bellow or something else ?
> libpjsua.so (libc6) => /usr/lib/libpjsua.so
>
You will likely need to pass the pjproject directory to configure.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github
On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen
wrote:
> Hello,
>
> Thank You Paul for your reply,
>
> The registrations in my setup are not duplicated, the 'secret' field in the
> realtime table is empty, which causes Asterisk to not authenticate requests
> fro
is needed for asterisk (in fact 1.8 has no
support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a
kamailio peer and away you go.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https:/
another 20-30 minutes. It seems that there is a
poll interval of some kind. I am not sure whether it is a setting on the phone
or the asterisk.
Any ideas?
Best Regards,
Paul Greenberg, Esq.
Law Office of Paul Greenberg
530 Main Street, Suite 102
Fort Lee, NJ 07024
E-mail: p...@greenberg.pro
Tel
can reduce the window of opportunity for that by several 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
On Thu, Apr 17, 2014 at 7:25 AM, binary dreamer
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.
--
Paul Belanger | PolyBeacon,
he
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.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitt
27; before handing off to Asterisk -- easier to
> implement, easier to maintain, no legal BS to consider.
>
> 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) ?
ge you 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.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https:
Madsen on this and he's updating the site now. Currently
only the 3rd edition is published online.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger
en a user changes their password, secret.conf gets
updated not voicemail.conf.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/p
> bandwidth), but I'd lose some functionality and have to re-write parts of my
> application.
>
> Any clues of what limit I'm hitting and how to increase it?
>
DAHDI has a pseudo channel limit of 512, some
Generator: 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 voicem
ss?
>
What sort of channel count 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.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
OpenStack. So you could offer your customers a self-managed, redundant
> Asterisk cloud or something like that. :)
>
> In theory, this combination should give you a 100% redundant, auto-healing,
> auto-scaling VoIP setup. :)
>
+1 to this post. A lot of good information here.
primary box 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.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenod
On 05/03/14 12:56, Paul Hayes wrote:
I appreciate that and I do understand why but that setting doesn't work
as described, it seems to do nothing.
While we're at it, what's the recommended alternative method to replace
using "asterisk -rx" in bash scripts now?
cheer
While we're at it, what's the recommended alternative method to replace
using "asterisk -rx" in bash scripts now?
cheers,
Paul.
Jerry
--
David M. Lee
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: w
On Thu, Feb 27, 2014 at 10:55 PM, Darryl Moore wrote:
>
> On Feb 27, 2014 10:02 PM, "Paul Belanger"
> wrote:
>>
>> >
>> No such thing as 'free open source g729 license', if you actually read the
>> site:
>>
>
> There is regardin
s reluctant to bring this topic up yet again , and yes I did
> google around and read the different material on the subject however,
> I am still in need of some definitive answers.
>
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pab
;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285
>
> Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
>
> From: "Haley, Scott"
> ;tag=8066eb6f589ce3124b652973b4b00
>
> To: ;tag=as06e2e068
>
> Call-ID: 8066eb
will fail to establish because lack of
codecs. If you offer a both g729 and ulaw, then ulaw will be used.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.c
What you describe is more 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.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freeno
On Thu, Feb 13, 2014 at 1:04 AM, George Joseph
wrote:
> On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger
> wrote:
>>
>> On Wed, Feb 12, 2014 at 12:50 PM, Olivier wrote:
>> > Hello,
>> >
>> > How does extensions.lua compares to extensions.conf or exte
ing AGI if you want to leverage redis or memcached.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
__
ing "au" for now.
Is there any way to reduce the startup latency and make MixMonitor
write the audio stream to the output file immediately? I looked
briefly at apps/app_mixmonitor.c and main/file.c but I don't fully
understand the code. Is mixmonitor forking an external conversio
ty for Asterisk. If you are using
SIP, you want to REINVITE media 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/
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@
:5] Wait("SIP/abcde-0016", "5") in
> new stack
> -- Executing [12345678912@from-sip:6] Dial("SIP/abcde-0016",
> "SIP/123&SIP/456,30,oxX") in new stack
> == Using SIP RTP CoS mark 5
> == Using SIP RTP CoS mark 5
>
,
>
> When we have bindport = 172.x.x.14 then "netstat -udpln" shows the
> following. When bindport is 0.0.0.0 then netstat shows it listening on
> 0.0.0.0 as you'd expect.
>
> udp0 0 172.x.x.14:50600.0.0.0:*
> 18114/asterisk
>
>
> --
On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
wrote:
> On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
> wrote:
>> Hi Paul,
>>
>> Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
>> and arriving at the Asterisk server. This is why i
On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
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 Mon, Jan 20, 2014 at 4:24 PM, David Cunningham
wrote:
> Hi Paul,
>
> The ngrep on the Asterisk server does show it being received. Have you any
> idea what would prevent it getting from the network stack to Asterisk on
> that machine?
>
Well, you need to use tcpdump on ea
erver's real address
> 192.z.z.z is the calling phone's LAN address
>
Sounds like a routing problem opposed to an application issue. You'll
have to fire up tcpdump on Kamailio and see what happens to the
packet. The look at the local routing tables to see where it is
getting
FRAMEWORK is an option selectable under the "Compiler Flags -
Development" menu in menuselect.
./configure --enable-dev-mode
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https
your kind response!
Save yourself time / energy 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.
hts ?
>
> Regards
>
I basically had the same issue as you for one of my sites. I tried
everything under the sun to figure it out, change cables, loop back
test, change out hardware, clocking, etc.
In the end I had to upgrade dahdi to 2.7+ and the issue
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.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: p
60
> negative_connection_cache => 600
>
>
> -- /etc/asterisk/cdr_adaptive_odbc.conf lists below:
> [cdr]
> connection=asterisk
> table=cdr
> alias start => calldate
> alias phoneno => phoneno
> alias userid => userid
> alias callerid => c
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
On Dec 17, 2013, at 1:29 AM, virendra bhati wrote:
> Good Paul,
>
> I used Konference a lot very nice apps, but will this work with asterisk
> latest version or not ?
>
It should work on the latest asterisk version.
> I used asterisk 1.4,1.8 but didn't work on 11...
mixed and whispered to the spyee.
--
Paul Albrecht
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org
der adding the OTHER 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 Belang
are saying?
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
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?
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan
gt; _X,1,NoOp(Digit entered during prompt)
exten => _X,2,Goto(project,s,1)
Then you have a DTMF issue, Background will allow DTMF to interrupt the
prompts.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger |
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?
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github
to(project,s,1)
exten => i,1,Playback(${sounds_path}error)
exten => i,n,goto(project,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 spee
700 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
Then make an educated guess about what is
happening.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabel
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.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger
ver, solve 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.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitt
d and then breaking
all of those down by customer as we run a multi tenanted set up.
SNMP would give us totals but I don't think it would do the breakdown by
customer.
You should avoid using the CLI to access that information. You'd likely
getter better results using AMI or CEL.
--
Paul B
need to change to a different one.
Both flowroute.com and voip.ms work well 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 c
nformation.
[1] https://wiki.asterisk.org/wiki/display/AST/AMI+1.4+Specification
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
--
1 - 100 of 2972 matches
Mail list logo
