[asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

Bruce B bruceb444 at gmail.com
Sat Jul 23 08:49:56 CDT 2011


Not really. It's only good after DECLINED is sent.

On Sat, Jul 23, 2011 at 2:08 AM, Mitesh Thakkar <mail.mthakkar at gmail.com>wrote:

> I think fail2ban can help in this issue.
>
> Regards,
> Mitesh Thakkar
> +91 94279 07952
> Yahoo: miteshthakkar_nu at yahoo.co.in
> GTalk: mail.mthakkar at gmail.com
>
>
>
> On Sat, Jul 23, 2011 at 10:04 AM, Bruce B <bruceb444 at gmail.com> wrote:
> > Robert thanks for weighing in.
> > So, you are saying that FreeSwitch on it's own can tackle issues like
> this
> > without the need of OpenSIPs? Can you elaborate please?
> > Thanks
> >
> > On Sat, Jul 23, 2011 at 12:17 AM, Robert-iPhone <rhuddleston at gmail.com>
> > wrote:
> >>
> >> I like to put mine on 3389
> >>
> >> hahaha just kidding.
> >>
> >> Personally I'm starting to convert to FreeSwitch - oops I had to say it.
> >>
> >> Security can be difficult and there are some good SBCs out there - just
> >> begs investment in technology - OH and bright staff
> >>
> >>
> >> Sent from my iPhone
> >>
> >> On Jul 23, 2011, at 12:09 AM, Steve Edwards <asterisk.org at sedwards.com>
> >> wrote:
> >>
> >> > On Fri, 22 Jul 2011, Bruce B wrote:
> >> >
> >> >> 1- So, you are saying that either of OpenSER/Kamailio/OpenSIPS
> actually
> >> >> give me the full capability to the SIP stack to do the sort of thing
> I was
> >> >> asking for? And this can run on the same server as Asterisk is
> running?
> >> >
> >> > Configure OpenSIPS to listen to 5060 and Asterisk to listen to 5061.
> >> >
> >> > --
> >> > Thanks in advance,
> >> >
> >> >
> -------------------------------------------------------------------------
> >> > Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867
> >> > PST
> >> > Newline                                              Fax:
> >> > +1-760-731-3000
> >> > --
> >> > _____________________________________________________________________
> >> > -- 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 UNSUBSCRIBE or update options visit:
> >> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >> --
> >> _____________________________________________________________________
> >> -- 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 UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > _____________________________________________________________________
> > -- 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 UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --
> _____________________________________________________________________
> -- 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 UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110723/e76f453b/attachment.htm>


More information about the asterisk-users mailing list