On Sat, 15 Oct 2016, tux john wrote:
Hi. Kinda new to the area and I would like some help please. I have
asterisk 11 in my system and I have 10 users and 12 DIDs. One did routed
to each user and 2 DIDs for faxing. Everything works fine but I do not
have call transfer between extensions and feature access codes. I have
read somewhere that enabling call transfer can be a security hole for
sip attackers.
Are these incoming calls copper or VOIP?
If you only accept copper calls, make sure Asterisk is only listening to
127.0.0.1 and enforce this policy with another layer dropping any incoming
SIP packets at the firewall.
If you only intend to accept calls from your ISP, configure Asterisk to
only accept calls from your ISP, and enforce this policy at the firewall.
If you accept calls from everyone, re-think your definition of 'everyone.'
It probably does not include Iraq, North Korea, China, Russia, etc.
Configure Asterisk and your firewall accordingly.
Beyond this, follow 'best practices' (google for sip best practices --
John Todd did a list years back, Nerdvittles probably will also be a good
resource) like long, random user names and passwords, only allow needed
features to each class of users, etc.
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards [email protected] Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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