[asterisk-users] How to configure asterisk to only accept SIP from kamailio at localhost but exchange RTP on all interfaces?
Alex Villacís Lasso
a_villacis at palosanto.com
Tue Feb 25 12:04:55 CST 2014
El 25/02/14 08:30, Karsten Wemheuer escribió:
> 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=127.0.0.1 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
>> 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
> If You bind asterisk to 127.0.0.1 I think the media connection is set
> for this IP. Your Softphone can not reach the correct 127.0.0.1
> (localhost is everywhere).
> I would suggest, You setup asterisk on eth0 address or 0.0.0.0. In the
> sip.conf You could secure Your setup with
> deny = 0.0.0.0/0.0.0.0
> permit = Your-LAN-Adress
> This way asterisk accepts SIP from Your box only.
This might work, but would need to touch sip.conf every time the IP address changes. It would be nice to have a configuration that can be set up once and not modified again. That is why I wanted to set up localhost.
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