Hi everyone, I've spent the day debugging NAT (BTW, I am fully ipv6 ready... hint hint), and I'm heading for insanity. I call into the phone system with my cell, and dial my extension at the AA. When I pick up on my Polycom phone, I can hear the voice from the cell phone. However, the cell phone has no incoming audio (no sound through speaker) at all (but I did get the message from the AA). Everything works extension-to-extension, so I suspect a NAT issue. A bit of background... I have a T1, and the firewall that also does NAT has a static IP.
I have followed a number of guides online, and tried to cobble together what to do. Here is the general layout of my firewall rules: - Allow port 5060 through to SipX (both TCP and UDP) - Allow port 5080 through to SipX (both TCP and UDP) - Allow port range 30000-310000 through to SipX (UDP only) Outgoing traffic is always accepted. I have installed nt_conntrack_sip and nf_nat_sip. They seem to work, since I disabled the rules for port-forwarding UDP 30000-31000 and things still work. I indicated to voicnetwork that it should connect via port 5060, and set that I am behind a NAT (it has some setting for this, which I do not fully understand). On the SipX side, I have done the following: System->Internet Calling->NAT Traversal: Both "Enable NAT Traversal" and "Server Behind NAT" are enabled System->Servers->xxx->NAT: "Address Type" is "Specify IP Address"... and the public IP address for the firewall which does NAT is provided... SIP port is 5060... A small bit of background... this is the only SIP trunk that will be configured, and it does incoming and outgoing calls. All phones connected to the system are behind the NAT. Any advice would be greatly appreciated... AJ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
