Hi all,
Finally we make decision that go with PortSIP, the reasons are below:
1. Support the easy cluster deployment for handle large concurrent calls
and provide
2. All REST API(this is very important to us for integrate the PBX with our
current system), and also offer the rebrand app for free.
On Thu, Apr 20, 2017 at 05:51:59PM -0300, Fabio Moretti wrote:
> Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
> >
> > This gets kinda Rube Golberg-ish, but convert the incoming analog line
> > to sip, route it through asterisk and have asterisk do its thing
> > before
On Thursday 20 Apr 2017, Fabio Moretti wrote:
> Hi,
>
> I've some analogic lines and I'm asked if it's possible to program an
> asterisk for "checking" the inbound calls without answering them, doing
> something like this:
>
> analog line 1 -+-- asterisk
>
>
JFYI - https://issues.asterisk.org/jira/browse/ASTERISK-26922
чт, 20 апр. 2017 г. в 11:38, Kseniya Blashchuk :
> Hi!
> The issue did not reproduce with pjsip. As for ppa - somebody recommended
> me ppa:sapian/asterisk. Does anybody use it maybe?
>
>
> вт, 18 апр. 2017 г. в
Ah, ok.
Everytime you install a package you need to run configure again to
allow detection of new lib.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
Thank for all the replies, a lot of input and information!
Sorry for this useless mail, but I really wanted to say thank you.
Il 20/04/2017 17:26, Fabio Moretti ha scritto:
> Hi,
>
> I've some analogic lines and I'm asked if it's possible to program an
> asterisk for "checking" the inbound
Hello
you mean while placing a video call ? What info am I looking for in the
debug output ?
Kind regards.
J.
On 21-04-17 12:28, Marcelo Terres wrote:
Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?
Regards,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, April 21, 2017 10:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk
Hi, Jerry,
I don't know what S.O. you have in the Server, but you can check the man
page (https://linux.die.net/man/8/in.tftpd) for tftpd and use the options
--address, so you can tell tftp from what interface/port this service
listen request.
>From the IP in your logs (69.64.57.18) the request
Is that IP in your network or outside (I can ping it so I'm guessing it's
outside your network)? Do you have a firewall between your asterisk box
and the internet? Is there a WHITELIST of IP addresses that only allow
your provider's limited IP pool to connect to your asterisk box from
outside?
Hi David, Tim,
Try to use Bail2Ban at last resort. Fail2Ban is a ractive approach, that
permit the traffinc AND ONLY BLOCK them after certain level triggered.
Use iptables to block the unused services faced to public networks like
Internet. And configure these services properly, so they listen
Jonathan,
Have you tried:
same => n,Set(TIMEOUT(absolute)=3540)
You could override later if required.
Thanks,
Dan
On 17 Apr 2017 04:49, "Jonathan H" wrote:
The following setup prevents callers from going over 59 minutes:
--
the output of ls -l is
root@pbx: ~ $ ls -l /var/run/asterisk/asterisk.ctl
srwxr-xr-x 1 asterisk asterisk 0 Apr 20 19:47 /var/run/asterisk/asterisk.ctl
root@pbx: ~ $
On Thu, Apr 20, 2017 at 7:46 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> On Thursday 20 April 2017 at
I "justed" happened to look at /var/log/messages...
I saw:
Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename
0004f2034f6b.cfg
Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found
0004f2034f6b.cfg
Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Friday, April 21, 2017 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Hack attempt
This is old news. They use Shodan and then try to connect. Set up Fail2Ban
that say after 10 404's to ban the IP.
On Fri, Apr 21, 2017 at 12:27 PM, Jerry Geis wrote:
> I "justed" happened to look at /var/log/messages...
>
> I saw:
> Apr 21 12:18:40 in.tftpd[22719]: RRQ
You can use func_odbc to do this.
https://wiki.asterisk.org/wiki/display/AST/Getting+Asterisk+Connected+to+MySQL+via+ODBC2
There is a good chapter in the Asterisk book about using ODBC for
hotdesking that may help you understand ODBC as well.
Hello,
Best practices examples to upgrade Asterisk 13.13.1 to latest version?
Any suggestions?
Thanks,
Motty
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Check out the new Asterisk
hi. currently i am running the phonebook in astdb with
*database put cidname 0123456789 "name_surname"*
and i retrive it with
*exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})*
Now, my system has mysql and i got all my contacts in there in a database
is called *asterisk
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