Hi all,
after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show
what's happens. I've trying setting debug and verbose to 100 but
nothing, no show. All commands works as expected but i can't what's
happens on my asterisk server.
asterisk*CLI> core show settings
PBX Core settings
Can anyone comment on using SMS in conjunction with VoIP service using
one of these three VoIP providers: voip.ms, vitelity.com,
flowroute.com? Are some SMS services more compatible with Asterisk
(i.e. SMS over SIP works perfectly or not)? Is it best to use a
different data channel for SMS
terisk/outgoing/ and the SMS will be
sent.
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Brandon B.
Skickat: den 29 november 2016 17:25
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] Asterisk compatibility with SMS services
Can
Can anyone comment on using SMS in conjunction with VoIP service using
one of these three VoIP providers: voip.ms, vitelity.com, flowroute.com?
Are some SMS services more compatible with Asterisk (i.e. SMS over SIP
works perfectly or not)? Is it best to use a different data channel for
SMS
Hey Chris,
Starts from here,
https://wiki.asterisk.org/wiki/display/AST/Getting+Started or try Asterisk
Complete guide in pdf format. If you are looking for something graphical,
go for elastix or freepbx.
Thanks
~Arun
On Thu, Nov 24, 2016 at 12:28 AM, christopher kamutumwa <
Goodday users
Am quite new to asterisk and trying to configure it with an fxo and fxs
digium card. also i need a gui interface implemented. I have a centos 6.8
server any tutorial i could use for install and configuration? would
appreciate.
Thanks
Chris
--
With the release of Asterisk 11.25.0 [1] the Asterisk 11 branch has entered
into a security only maintenance phase (approximately 1 year).
What this means is that there will be no more general updates (bug fixes,
enhancements) and subsequent releases *except* in the case of a security
The Asterisk Development Team has announced the release of Asterisk 14.2.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.2.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 13.13.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.13.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 11.25.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.25.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 14.2.0-rc2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.2.0-rc2 resolves an issue reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 13.13.0-rc2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.13.0-rc2 resolves an issue reported by the
community and would have not been
On 21-11-16 17:20, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
> wrote:
On 21-11-16 15:17, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
On 21-11-16 19:14, Jonas Kellens wrote:
On 21-11-16 17:20, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
> wrote:
On 21-11-16 15:17, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 7:05 AM, Jonas
Hello,
>From [1], I read the following example:
[applicationmap]
retrieveinfo => #8
,peer,Set(ARRAY(CDR(mark),CDR(name))=${ODBC_FOO(${CALLERID(num)})})
Then I wrote is my own simplified example:
[applicationmap]
setuserfield => 11,peer,Set(CDR(userfield)=FOO)
When running the above example,
On 21-11-16 17:20, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
> wrote:
On 21-11-16 15:17, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens
wrote:
> On 21-11-16 15:17, Matthew Jordan wrote:
>
>
> On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
> wrote:
>
>> Hello
>>
>> when using Asterisk version 13.12.2 I notice that it takes up to
On 21-11-16 15:17, Matthew Jordan wrote:
On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
> wrote:
Hello
when using Asterisk version 13.12.2 I notice that it takes up to
30 seconds (sometimes even longer) for a call queue
On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
wrote:
> Hello
>
> when using Asterisk version 13.12.2 I notice that it takes up to 30
> seconds (sometimes even longer) for a call queue to call its members.
>
> Example 1 :
>
> [Nov 21 08:17:57] pbx.c: Executing
Hello
when using Asterisk version 13.12.2 I notice that it takes up to 30
seconds (sometimes even longer) for a call queue to call its members.
Example 1 :
[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack
[Nov 21
On a newly installed idle 11.13.1 enabled system, I do also have:
CLI> module show like timing
Module Description Use
Count
res_timing_pthread.so pthread Timing Interface
0
res_timing_timerfd.so Timerfd Timing Interface
0
2
I found this comment:
2014-04-08 21:20 + [r411944-411974] Richard Mudgett <
rmudg...@digium.com>
* main/asterisk.c, /: Internal timing: Add notice that the -I and
internal_timing option are no longer needed. Add notice messages
during execution that the -I
Hi,
Am 17.11.2016 um 13:51 schrieb Jerry Geis:
> PBX Core settings
> -
> Version: 11.24.1
> Build Options: LOADABLE_MODULES, BUILD_NATIVE
> Maximum calls: Not set
> Maximum open file handles: 1024
> Root console
module show like timing
Module Description Use
Count Status Support Level
res_timing_dahdi.soDAHDI Timing Interface 0
Running core
res_timing_pthread.so pthread Timing Interface
ssion
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Asterisk 11.24.1 garbled audio
> Message-ID:
> <CAPeT9jiQeM5BJJQXtstKyA6k3GLbDSN45Z8WYLaSu24UwAzfEQ@mail.
> gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
>
Date: Tue, 15 Nov 2016 17:52:07 +0100
From: Olivier <oza.4...@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Asterisk 11.24.1 garbled audi
Hi,
Am 15.11.2016 um 17:52 schrieb Olivier:
> Hi,
>
> How can I double check which timer is currently is use in a running system ?
> core show settings doesn't tell anything, if I'm not mistaken.
To determine which timing module is currently in use, you can take a look at
"module show like
Is it possible to create a call file on asterisk that would call a video
phone and play a video file ?
What is the format of that file?
I can do this now with a softphone for audio and it works fine.
I am familiar with audio call files and use them all the time.
Video is now a different
Hi,
How can I double check which timer is currently is use in a running system ?
core show settings doesn't tell anything, if I'm not mistaken.
Regards
2016-11-11 21:02 GMT+01:00 Matthew Jordan :
>
>
> On Fri, Nov 11, 2016 at 10:46 AM, Jerry Geis
On 11-10-16 14:44, Joshua Colp wrote:
Jonas Kellens wrote:
Hello
I am experiencing a freeze of the Asterisk proces when issuing a 'sip
reload'.
I have this issue every time on asterisk versions : 13.11.2, 13.11.1,
13.10.0 and certified-13.8-cert3.
I do not have this on versions
On Fri, Nov 11, 2016 at 10:46 AM, Jerry Geis wrote:
> >Information on timing sources can be found here:
>
> >https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
>
> >As noted on that page, ConfBridge can use any timing interface Asterisk
> >provides, and is not
>Information on timing sources can be found here:
>https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
>As noted on that page, ConfBridge can use any timing interface Asterisk
>provides, and is not limited to the DAHDI timing interface. Generally,
>timerfd is a good timing interface.
On Thu, Nov 10, 2016 at 4:00 PM, Jerry Geis wrote:
> I found dahdi_test...
>
> dahdi_test
> Opened pseudo dahdi interface, measuring accuracy...
> 99.999% 99.904% 99.974% 99.814% 98.070% 97.850% 99.985% 99.887%
> 99.708% 99.899% 99.805% 99.708% 99.902% 100.000% 99.949%
I found dahdi_test...
dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.999% 99.904% 99.974% 99.814% 98.070% 97.850% 99.985% 99.887%
99.708% 99.899% 99.805% 99.708% 99.902% 100.000% 99.949% 99.883%
99.891% 99.906% 99.784% 99.719% 99.827% 99.903%
--- Results after 22 passes ---
Hi all
I am using asterisk 11.24.1 on a centos 5 machine. kernel 2.6.18 flavor.
(x86_64).
I have about SIP 150 endpoints on it.
when I send a message I'm getting garbled audio.
I used to have a single PRI card in the box - but something happened and
that connection
no longer worked. I removed
The Asterisk Development Team has announced the release of Asterisk 14.1.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.1.2 resolves an issue reported by the
community and would have not been possible without
The Asterisk Development Team has announced the release of Asterisk 13.12.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.12.2 resolves an issue reported by the
community and would have not been possible
Does anyone know if Asterisk 13 will support T.38 Version 3?
?
Thanks
Bryant
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at:
On Mon, Nov 7, 2016, at 02:12 PM, Motty Cruz wrote:
> It does not append the area code:
> 5007-0047","SIP/truck1-0048","Dial","SIP/truck1/6052736,80","2016-11
> -06 18:49:41",,"2016-11-06 18:49:45",4,0,"NO
> ANSWER","DOCUMENTATION","1478458181.738",""
>
004c","Dial","SIP/truck1/6052736,80","2016-11
-06 18:54:02",,"2016-11-06 18:54:23",20,0,"NO
ANSWER","DOCUMENTATION","1478458442.914",""
Our Sip provider takes 10 digits, it should have append 381 to 6052736
numb
On Sun, Nov 6, 2016, at 03:35 PM, Motty Cruz wrote:
> Hello, I would like to add area code to local numbers, it worked like a
> charm on Asterisk 1.8 but does not work on Asterisk 13.11.
>
>
>
> Extensions.conf; worked before on Asterisk 1.8
> ; Adding Area code to local numbers
>
> exten =>
Hello, I would like to add area code to local numbers, it worked like a
charm on Asterisk 1.8 but does not work on Asterisk 13.11.
Extensions.conf; worked before on Asterisk 1.8
; Adding Area code to local numbers
exten => _9XXX,n,Set(CALLERID(all)="$CallerID" <3818008000>)
exten =>
The Asterisk Development Team has announced the release of Asterisk 13.12.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.12.1 resolves an issue reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 11.24.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.24.1 resolves an issue reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 14.1.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.1.1 resolves an issue reported by the
community and would have not been possible without
>>> On Oct 26, 2016, at 8:20 AM, Joshua Colp jc...@digium.com wrote:
>>> applicable to those two operations. If you create an issue I can throw a
>>> patch up.
Done:
https://issues.asterisk.org/jira/browse/ASTERISK-26503
Doug
--
Doug Lytle wrote:
On Oct 25, 2016, at 5:10 PM, Asterisk Development Team asteriskt...@digium.com
wrote:
The Asterisk Development Team has announced the release of Asterisk 11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
Just
>>> On Oct 25, 2016, at 5:10 PM, Asterisk Development Team
>>> asteriskt...@digium.com wrote:
>>> The Asterisk Development Team has announced the release of Asterisk 11.24.0.
>>> This release is available for immediate download at
>>> http://downloads.asterisk.org/pub/telephony/asterisk
Just
Le 26/10/2016 à 12:21, Joshua Colp a écrit :
Administrator TOOTAI wrote:
Le 25/10/2016 à 23:10, Asterisk Development Team a écrit :
The Asterisk Development Team has announced the release of Asterisk
11.24.0.
This release is available for immediate download at
Administrator TOOTAI wrote:
Le 25/10/2016 à 23:10, Asterisk Development Team a écrit :
The Asterisk Development Team has announced the release of Asterisk
11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk
Le 25/10/2016 à 23:10, Asterisk Development Team a écrit :
The Asterisk Development Team has announced the release of Asterisk 11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.24.0 resolves several
Hey All,
This is a friendly notice that as of today Asterisk 11 has entered
security fix only mode. From this point onward additional releases of
Asterisk 11 will no longer be made unless there is a security fix
being applied to the branch. Users of Asterisk 11 are encouraged to
move to one of
The Asterisk Development Team has announced the release of Asterisk 14.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.1.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 13.12.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.12.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.24.0 resolves several issues reported by the
community and would have not been possible
Thank you for your help! Centos 7 firewall was enable.
systemctl stop firewalld
issue fixed.
Thanks,
On Thu, Oct 13, 2016 at 3:54 PM, Victor Villarreal
wrote:
> Ok.
>
> Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of
> the Polycom hardphone.
On Thu, Oct 13, 2016 at 12:06 PM, wrote:
> > I have Asterisk running well inside our network. I did some
> > experiments exposing it to internet but had some issues:
> > 1. NAT issues (voice one way, etc). From what I understand double-
> > NAT users will always
A few years ago I ran into something similar. Using TLS seemed to fix
it, but it was a while ago so I might be wrong.
On 10/14/2016 11:35 AM, Greg Woods wrote:
On Fri, Oct 14, 2016 at 9:06 AM, Dovid Bender > wrote:
Changing your port
On Fri, Oct 14, 2016 at 9:06 AM, Dovid Bender wrote:
> Changing your port should fix all your worries.
>
That may work if you control both ends of the SIP connection.
--Greg
--
_
-- Bandwidth and
On Fri, Oct 14, 2016 at 7:55 AM, Jerry Geis wrote:
> Apparently Verizon is blocking or changing packets on port 5060 so my
> softphone from my hotspot will not work.
>
Sounds like you are another victim of SIP ALG. I ended up having to change
to a VOIP provider that would
Changing your port should fix all your worries.
On Fri, Oct 14, 2016 at 11:00 AM, Greg Woods wrote:
>
>
> On Fri, Oct 14, 2016 at 7:55 AM, Jerry Geis wrote:
>
>> Apparently Verizon is blocking or changing packets on port 5060 so my
>> softphone from
iptables -t nat -A PREROUTING -i inboundinterface -s sourceip -d destinationip
-p udp --dport 5070 -j DNAT --to destinationip:5060
Or something similar. You may however found that the provider's filtering is
application based rather than port based.
--
Sent from my cellphone.
--
They don't like competition ;)
On Fri, Oct 14, 2016 at 9:55 AM, Jerry Geis wrote:
> Apparently Verizon is blocking or changing packets on port 5060 so my
> softphone from my hotspot will not work.
>
> How do I set asterisk (11.23.0) to run default 5060 for all other
Apparently Verizon is blocking or changing packets on port 5060 so my
softphone from my hotspot will not work.
How do I set asterisk (11.23.0) to run default 5060 for all other devices I
have - BUT for my software run on a different port like 5070? I'm using
linphone and is easy to change the
On Thu, Oct 13, 2016 at 12:35 PM, Brandon B. wrote:
> What part of Asterisk 14.0.2 opens the random, high numbered (33094
> currently) UDP port? This port is opened even without any channel drivers
> loaded.
>
> $ sudo netstat -ltunp | grep asterisk
> udp0 0
Ok.
Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of
the Polycom hardphone. If this is true, then you have NAT issues.
The REGISTER message are received by your PBX, but when respond, Asterisk
send the next SIP message to the IP informed by the phone, that is the
on public IP, client is on private
network.
Thanks,
Motty
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Victor Villarreal
Sent: Thursday, October 13, 2016 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
S
What part of Asterisk 14.0.2 opens the random, high numbered (33094
currently) UDP port? This port is opened even without any channel
drivers loaded.
$ sudo netstat -ltunp | grep asterisk
udp0 0 0.0.0.0:51488
0.0.0.0:* 13830/asterisk
udp0
Hi Motty,
Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then "sip set
debug on".
Now try to register again. At last, " sip de debug off".
Examine tour console or full log file to find some clue ir send me back
some trace.
Cheers.
El oct. 13, 2016 1:45 PM, "Motty Cruz"
I think you had asked what phone works well with VPN's. I've had very
good experiences with Yealink using OpenVPN, never an issue.
I think I've heard that Snom does OpenVPN as well.
Mark
--
_
-- Bandwidth and Colocation
Hello, fresh install of Asterisk 13.11.2, client unable to register. For
now I have IPtables disabled, also selinux is disabled
[1006]
type=friend
username=1006
secret=mysecret
context=sip-phone
call-limit=1
callerid="iuser" <1006>
disallow=all
host=dynamic
allow=all
any ideas?
> I have Asterisk running well inside our network. I did some
> experiments exposing it to internet but had some issues:
> 1. NAT issues (voice one way, etc). From what I understand double-
> NAT users will always have something like this
> 2. Immediately I see people trying to hack into. I did
Hello list,
I have Asterisk running well inside our network. I did some experiments
exposing it to internet but had some issues:
1. NAT issues (voice one way, etc). From what I understand double-NAT users
will always have something like this
2. Immediately I see people trying to hack into. I
Are those numbers correct?
Asterisk 12 stopped being supported almost 2 years ago and became "do
not use" on 2015-12-20
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Ubuntu 14 may still be supported, if you're on 14.0.4.5
https://wiki.ubuntu.com/Releases
You could try make
Hello,
Am trying to install asterisk 12 on ubuntu 14.04lts and am getting the
below error after a MAKE any hints how to go round this?
bedit.a -> asterisk
asterisk.o: file not recognized: File truncated
collect2: error: ld returned 1 exit status
make[1]: *** [asterisk] Error 1
make: *** [main]
Hi Jonas!
Do you currently use any TLS technology in your Asterisk? Like SIP-TLS o
pjSIP-TLS support ? If don't, please go to modules.conf and start disabling
some modules that you don't use.
For example, I can see some other modules related to calendars. If you
don't use this, please disable
ailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
*Sent:* Tuesday, October 11, 2016 8:41 AM
*Subject:* [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and
certified-13.8-cert3 : freeze on 'sip reload'
Hello
I am experiencing a freeze of the Asterisk proces
ent: Tuesday, October 11, 2016 8:41 AM
Subject: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and
certified-13.8-cert3 : freeze on 'sip reload'
Hello
I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'.
I have this issue every time on asterisk versio
Jonas Kellens wrote:
Hello
I am experiencing a freeze of the Asterisk proces when issuing a 'sip
reload'.
I have this issue every time on asterisk versions : 13.11.2, 13.11.1,
13.10.0 and certified-13.8-cert3.
I do not have this on versions certified-13.8-cert2,
certified-13.8-cert1 and
Hello
I am experiencing a freeze of the Asterisk proces when issuing a 'sip
reload'.
I have this issue every time on asterisk versions : 13.11.2, 13.11.1,
13.10.0 and certified-13.8-cert3.
I do not have this on versions certified-13.8-cert2,
certified-13.8-cert1 and asterisk 1.8.32.3.
Hi all ! Thanks for your feedback and sory for the delay. Respond:
> Date: Mon, 3 Oct 2016 21:05:55 -0300
> From: Marcelo Terres
>
> I think that you need the dev files too. In Debian 8, the package is
> libmysqlclient-dev.
>
> But Debian 8 uses libmysqlclient-18. Where did
and set permissions 777 for radiusclient configs,
> but the issue remains the same.
>
> Please advise the fix for resolving this issue.
>
>
>
> Date: Thu, 29 Sep 2016 18:11:15 +0800
>> From: Andrew Ivins <and...@ivins.id.au>
>> To: wil...@offermans.rompen.nl
>>
Jonathan H wrote:
Just a minor thing: on
http://www.asterisk.org/downloads/asterisk/all-asterisk-versions it
still reports 14.0.1 as being the latest version, although the download
itself contains 14.0.2
I'd have file a bug but there doesn't seem to be a "website" section in
the tracker.
I've
Just a minor thing: on
http://www.asterisk.org/downloads/asterisk/all-asterisk-versions it still
reports 14.0.1 as being the latest version, although the download itself
contains 14.0.2
I'd have file a bug but there doesn't seem to be a "website" section in the
tracker.
On 30 September 2016 at
On Mon, Oct 03, 2016 at 07:54:23PM -0300, Victor Villarreal wrote:
> Hi List!
>
> I'm facing a problem while compiling Asterisk-11 on a Debian 8 server.
>
> The mysql-server version installed is 5.7 and come from the official mySQL
> community repo for Debian.
For the record, we ubild both
I think that you need the dev files too. In Debian 8, the package is
libmysqlclient-dev.
But Debian 8 uses libmysqlclient-18. Where did you get the 20 ?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
Hi List!
I'm facing a problem while compiling Asterisk-11 on a Debian 8 server.
The mysql-server version installed is 5.7 and come from the official mySQL
community repo for Debian.
After compile, install and execute Asterisk, the comman "lsof -p `pidof
asterisk` | grep mysql" don't produce any
Hello.
This is the link of the slides of my talk presented yesterday in
AstriCon, about Asterisk and XMPP.
As soon as the video is available, I'll share it too.
http://pt.slideshare.net/mhterres/astricon-2016-using-asterisk-and-xmpp-to-provide-greater-tools-to-your-customers-and-your-users
[]s
The Asterisk Development Team has announced the release of Asterisk 14.0.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.0.2 resolves several issues reported by the
community and would have not been possible
hi,
i'm trying configure $subj
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger
but there is a ton of "informational" messages
[Sep 30 14:40:16] SECURITY[18311] res_security_log.c:
as able to send over to radius server without
any
> > > issue i.e. using same radiusclient config that I'm using for Asterisk
> > > radiusclient.
> > >
> > > Btw, will try to work on Andrew advise and will update you if I make
any
> > > progress.
> &
> >
> >
> > Date: Wed, 28 Sep 2016 10:09:51 +0200
> > > From: Willy Offermans <aster...@offermans.rompen.nl>
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > <asterisk-users@lists.digium.com>
> >
rmans.rompen.nl>
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > <asterisk-users@lists.digium.com>
> > Subject: Re: [asterisk-users] Asterisk Radius CDR
> > Message-ID: <20160928080951.ga4...@vpn.offrom.nl>
> > Content-Type: text/pl
progress.
Date: Wed, 28 Sep 2016 10:09:51 +0200
> From: Willy Offermans <aster...@offermans.rompen.nl>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Asterisk Radius CDR
> Mess
dius_log: Unable to create RADIUS record. CDR
> > not recorded!
> >
> > Please advise if I missed out anything.
> >
> >
> > Date: Mon, 26 Sep 2016 12:09:34 +0200
> >> From: Willy Offermans <aster...@offermans.rompen.nl>
> >> To: Asteris
: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users@lists.digium.com>
>> Subject: Re: [asterisk-users] Asterisk Radius CDR
>> Message-ID: <20160926100934.gb4...@vpn.offrom.nl>
>> Content-Type: text/plain; charset=us-ascii
>
While I agree with Nitesh that Nagios has some great monitoring tools, I
would recommend that you use Icinga 2 rather than Nagios with those
plugins,
Icinga has a bit more flexibility and better structure after it was forked
from Nagios.
We are using it to monitor everything from server status
The Asterisk Development Team has announced the release of Asterisk 14.0.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.0.1 resolves an issue reported by the
community and would have not been possible without
not recorded!
Please advise if I missed out anything.
Date: Mon, 26 Sep 2016 12:09:34 +0200
> From: Willy Offermans <aster...@offermans.rompen.nl>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-us
The Asterisk Development Team is pleased to announce the release of
Asterisk 14.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
Asterisk 14 is the next major release series of Asterisk. It is a Standard
Support release, similar to
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