On 20:00, Mon 13 Jun 05, Frank Cases wrote:
current setup
SIP phone 192.168.1.30 -- linksys wrt54g sveasoft -- INTERNET --
(xl0) Firewall (xl2:172.16.0.50)-- (em1:172.16.0.101) Asterisk
problem is RTP stream not oging trouhg from * to sip and vice versa.
#1 and asterusk is pushing
On Tuesday 14 June 2005 02:04, Michiel van Baak wrote:
On 20:00, Mon 13 Jun 05, Frank Cases wrote:
current setup
SIP phone 192.168.1.30 -- linksys wrt54g sveasoft -- INTERNET --
(xl0) Firewall (xl2:172.16.0.50)-- (em1:172.16.0.101) Asterisk
problem is RTP stream not oging trouhg
current setup
SIP phone 192.168.1.30 -- linksys wrt54g sveasoft -- INTERNET --
(xl0) Firewall (xl2:172.16.0.50)-- (em1:172.16.0.101) Asterisk
problem is RTP stream not oging trouhg from * to sip and vice versa.
#1 and asterusk is pushing 192.168.1.30 back to linksys with 172 as
return
I took the info from here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20firewall%20rules
and ended up with the following in my pf.conf:
rdr on $ext_if proto tcp from any to ($ext_if) port 5060 - $dmz_ip port 5060
rdr on $ext_if proto udp from any to ($ext_if) port 5060 - $dmz_ip port
