Hi,

I have tested with pfsense using the following NAT inbound rules (and
correspending filter rules):

WAN     TCP      5060    192.168.59.9    5060    pbx.dspnet.nl (SIP protocol 
oud)
WAN     TCP     11720   192.168.59.9    11720   pbx.dspnet.nl (H.323 Call Signal
Alternate)
WAN     TCP/UDP         1300    192.168.59.9    1300    pbx.dspnet.nl (H.323 
Hostcall secure)
WAN     TCP/UDP         1718    192.168.59.9    1718    pbx.dspnet.nl (H.323 
Gatekeeper
Discovery)
WAN     TCP/UDP         1719    192.168.59.9    1719    pbx.dspnet.nl (H.323 
Gatekeeper RAS)
WAN     TCP/UDP         1720    192.168.59.9    1720    pbx.dspnet.nl (H.323 
Hostcall)
WAN     TCP/UDP         1731    192.168.59.9    1731    pbx.dspnet.nl (H.323 
Audio call
control)
WAN     TCP/UDP         2979    192.168.59.9    2979    pbx.dspnet.nl (H.263 
Video Streaming)
WAN     TCP/UDP         3478    192.168.59.9    3478    pbx.dspnet.nl (STUN 
protocol)
WAN     TCP/UDP         8000 - 8011     192.168.59.9    8000 - 8011     
pbx.dspnet.nl
(RT(C)P Mediastream X-Lite Soft Phones)
WAN     UDP     1755    192.168.59.9    1755    pbx.dspnet.nl (Netshow)
WAN     UDP     2727    192.168.59.9    2727    pbx.dspnet.nl (MGP protocol)
WAN     UDP     4569    192.168.59.9    4569    pbx.dspnet.nl (IAX2 protocol)
WAN     UDP     5004    192.168.59.9    5004    pbx.dspnet.nl (RTP Stream)
WAN     UDP     5036    192.168.59.9    5036    pbx.dspnet.nl (IAX protocol)
WAN     UDP     5060 - 5082     19.168.59.9     5060 - 5082     pbx.dspnet.nl 
(SIP protocol)
WAN     UDP     9000 - 32767    192.168.59.9    9000 - 32767    pbx.dspnet.nl 
(RTP
Mediastream Phones)

However I noticed different phones are using different ports for the RTP
stream. Maybe it is necessary to adjust the used RTP stream ports in your
phone or Askozia, I am not sure about that.

With kind regards
Paul

> Hi all,
>
> We are using Commpartners as our VoIP provider.  I have set up a 1:1 NAT
> for an external IP which connects to our trunk at Commpartners.  I have
> the following Nat Forwarding rules set up:
>
> UDP 5060-5080
> UDP 10001-20000
>
> I can make calls outbound, but the problem is that from the VoIP phone I
> don't hear the standard ring tone when it connects to the phone number.
>  I used my cell phone as a test and I was able to call it, and from the
> VoIP phone I can talk, but i cannot hear myself talk from the cellphone
> to the VoIP phone, so it seems like it's only sending audio out, and not
> into the firewall.
>
> Has anyone else set up Tribox behind pfsense?  Looking for some guidance
> :)
>
> Thanks in advance.
>
> - Patrick
>
>


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