[asterisk-users] How to configure asterisk to only accept SIP from kamailio at localhost but exchange RTP on all interfaces?

Karsten Wemheuer kwem at gmx.de
Tue Feb 25 07:30:01 CST 2014

Hi Alex,

Am Donnerstag, den 20.02.2014, 13:48 -0500 schrieb Alex Villací­s Lasso:
> I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following
> the setup guide at
> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration 
> (MySQL database) so that kamailio authenticates and then forwards the
> registration to asterisk on localhost. The setup calls for asterisk to
> be configured to listen for SIP traffic on all interfaces, on a
> nonstandard port (I chose 5080). It also calls for 
> blanking of the password for the SIP peer (in my case, a softphone),
> so that it will not request for authentication again. I have managed
> to make a call with working audio from the softphone to an extension
> on asterisk through kamailio.
> My concern is that asterisk is left listening for SIP through all
> interfaces and with no SIP passwords. I want to secure the setup
> against directed traffic to the asterisk UDP port (5080), that
> bypasses the kamailio process. I tried setting 
> bindaddr= so asterisk will only listen for SIP traffic on
> localhost, but this has the side effect of also removing audio - the
> call appears to be successful on the softphone and on the asterisk
> logs, but no audio is actually heard. My theory is 
> that the RTP traffic is being sent to kamailio instead of the
> softphone.
> How can I set up asterisk so that it can send RTP anywhere but reject
> any SIP traffic that does not come from the kamailio process on
> localhost?

If You bind asterisk to I think the media connection is set
for this IP. Your Softphone can not reach the correct
(localhost is everywhere).

I would suggest, You setup asterisk on eth0 address or In the
sip.conf You could secure Your setup with
        deny =
        permit = Your-LAN-Adress
This way asterisk accepts SIP from Your box only.



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