Reg Sip Phones
I have Pfsense 1.2 RC3 installed in my firewall and i have one wan ( 1Mbps ) internet leased line.
I have Linksys , Polycom and Grandstream sip phones in my office.
When we configure these phones with firewall, the sip phones not getting incoming calls also the phone disconnected from Astrix server.
Please help me on this issue.
Search the forum for static-port.
It may be helpfull to tell asterisk what your local network is see the following link on the forum.
If you are using advanced outbound NAT you need to make sure to enable static-port on any subnet that has phones. More information on static outbound NAT is also available at the link above.
I need clear idea.
Please help me
If you want clearer directions then we have to know more details about your network.
Where is your asterisk server in relation to the phones? Are the phone in the same subnet as your asterisk server?
Are you using Asterisk directly or are you using Trixbox?
What steps have you taken so far to get this to work.
My setup is
i have 2 nos of Linksys PAP2 adaptors, 2 nos of Polycom, 4nos of Grandstream phones in my office.
My network id is 192.168.1.0
I have installed pfsense 1.2 in my system with Lan IP 192.168.1.1
One wan connection with static IP
My trixbox server located in remote place i mean not in our office.
i Have given pfsense ip as gateway for all phones.