You need to make sure the NAT type in your firewall is symmetric port nat. if your firewall is changing/randomizing the source OR destination port it will break audio.
You also need to make sure the itsp is sending calls to you on port 5080 and not 5060. If they send to you on 5060 you need to nat it to port 5080 if it is from the itsp's network on incoming traffic. On Mon, Oct 24, 2011 at 11:10 PM, Adrien Guillon <[email protected]>wrote: > 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/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services!
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