Some of us do allow sip from the internet, but just like for smtp email
protections are in order.
I point my SRV records at dedicated sip proxies (I use kamailio) which
check the INVITEd sip uri the same way my MXs check the SMTP Evelope-To
addresses, and only allow INVITEs through to authorized
On Thu, March 26, 2015 22:29, Michelle Dupuis wrote:
You have to consider whether you really want anonymous calls, or you
just want to enable SIP calls from trusted companies/partners. The
latter means setting up routes to these companies and (ideally)
registration between peers.
This is
James,
I'm a systems and telecom professional with experience going back more than
thirty years, to the days of teletype, current loop, POTS (2600hz signalling
anyone?) and echo
cancellation via analog level control and hybrid balance.
Your read of the intent of the VOIP/SIP design correctly.
On 27/3/15 8:03 pm, James B. Byrne wrote:
One only accepts VOIP calls from known correspondents. I
am not clear why this is so other than vague warnings respecting
(admittedly real and serious) security issues.
Because on the whole most people don't *want* to receive calls from
random
Dupuis
Cc: Asterisk Users List; byrn...@harte-lyne.ca
Subject: RE: [asterisk-users] Anonymous SIP calls
On Thu, March 26, 2015 22:29, Michelle Dupuis wrote:
You have to consider whether you really want anonymous calls, or you
just want to enable SIP calls from trusted companies/partners
Hi James,
Fortunately, your theory about common run for dollars is false with many
contra-examples. :)
jh
-- Původní zpráva --
Od: Bruce Ferrell bferr...@baywinds.org
Komu: asterisk-users@lists.digium.com
Datum: 28. 3. 2015 0:17:54
Předmět: Re: [asterisk-users] Anonymous
, March 26, 2015 9:24 PM
To: Asterisk Users List
Subject: [asterisk-users] Anonymous SIP calls
We have a FreePBX-12 / Asterisk-12 setup that supports about 24
extensions, most internal Snom870s but six or so external (Jitsi-2.8).
we use TLS and SRTP everywhere on our side of the fence. The server
We have a FreePBX-12 / Asterisk-12 setup that supports about 24
extensions, most internal Snom870s but six or so external (Jitsi-2.8).
we use TLS and SRTP everywhere on our side of the fence. The server
host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x)
and is up-to-date.
Hi all
Maybe somebody has an idea. I'm tracing a very strange phenomena...
I've a connection from Asterisk to a SIP PBX.
Most calls have a caller ID.
Some International calls don't have any.
Now it looks like those calls without caller ID never get to the context where
incomming calls from
