SIP One Way Audio

[marcocamza]

Hi All

I know this has been posting numerous times but I can't seem to get this right. I have pfsense 2.3.1-5 with 7 interfaces (vlans) everything is working as expected except for RTP.
I have a local asterisk pbx with phones on their own vlan for voice. I can register clients from external to pbx port 5060 and is successful but one way audio.
Have inbound port forwards set for 5060 and 10000:2000 UDP to Asterisk box (172.0.0.1). Outbound NAT set to Manual with static port.
When doing wireshark capture from external network static sip port matches Sip in registration. External IP(5060)<–->WAN IP(5060)<----->Internal IP172.0.0.1.
Upon SIP/SDP 1125 Request: INVITE sip:xxxxxxxxx@xxxxxxx.xxxxxx.com from external network

Owner/Creator, Session Id (o): - 1468936471315609 1 IN IP4 x.x.x.x (EXTERNAL IP IS CORRECT)

The return:
Owner/Creator, Session Id (o): root 2864 2864 IN IP4 172.0.0.1
Connection Information : IN IP4 172.0.0.1

PBX is maintained by voice providers but have been informed nat is setup on asterisk.
RTP audio will obviously be one way as it can't be route to private addresses but can't seem to figure if its the pbx or the firewall.

Anyone else experience this?

[chpalmer]

Quick note-  If you really are using 172.0.0.1 as your LAN you shouldn't be-  Unless your using AT&T uverse and have public IP addresses on your stuff.

Private space starts at 172.16.0.0

Lookup Result for 172.0.0.1

| IP Address: | 172.0.0.1 |
| Host of this IP: | 172-0-0-1.lightspeed.brhmal.sbcglobal.net |
| Organization: | AT&T U-verse |
| ISP/Hosting: | AT&T U-verse |