There are several things you should do: 1. Make sure there is a filter to *allow* all GRE traffic (protocol:any, not just tcp), on both systems.
2. I will also assume you have two different sip domains. I will also assume they can lookup and resolve each other SRV records and resolve them to a private ip address. If not, create a forward zone on each dns system to point to the other systemfor resolving that domain. 3. Create a gateway Enabled - yes name - gateway-for-300-range address - othersipdomain.com (whataver the name is of the sip domain of the OTHER system) 4. Create a "site-to-site" on system one to look like this: Enabled - yes name - dial-to-300-range Dialed Number prefix 3 and 2 digits Resulting Call - append entire dialed number gateway - gateway-for-300-range 5. Restart Services as prompted. You will be able to successfully dial from your 200 to your 300 range. When this happens you need to repeat the process on the 300 range to dial the 200 range. On Tue, May 18, 2010 at 1:04 AM, Rhon <[email protected]> wrote: > I'm using IPSEC GRE and pfsense interfaces have private IPs. should I still > need NAT for that matter? > > Thanks > > On Tue, May 18, 2010 at 3:03 AM, Picher, Michael <[email protected] > > wrote: > >> It should be set to manual and yes. >> >> >> >> *From:* Rhon [mailto:[email protected]] >> *Sent:* Monday, May 17, 2010 9:33 AM >> *To:* Picher, Michael; [email protected] >> *Subject:* Re: [sipx-users] No Voice/IVR on Site-to-Site >> >> >> >> Hello Michael, >> >> I have the static NAT port set to NO on pfsense. >> >> Also, to I have to enable NAT traversal on sipx? >> >> Thanks >> >> On Mon, May 17, 2010 at 3:20 PM, Picher, Michael < >> [email protected]> wrote: >> >> Static NAT port on the pfSense? >> >> >> >> *From:* [email protected] [mailto: >> [email protected]] *On Behalf Of *Rhon >> *Sent:* Monday, May 17, 2010 9:14 AM >> *To:* [email protected] >> *Subject:* [sipx-users] No Voice/IVR on Site-to-Site >> >> >> >> Hi, >> >> I have a problem with our deployment with SipXecs 4.2 which was installed >> fresh using ISO build. >> >> We cannot hear anything on both sides but are able to connect and can ring >> the other end. Calling the IVR is ok but no audio as well. >> >> SITE A: >> 100 - 199 >> >> SITE B: >> 200 - 299 >> >> Everything passed using Configurations tests. >> >> Our networks are setup as seen below: >> >> SITE A SIPX --> PFSENSE --> CISCO --> |||| VIA GRE TUNNEL |||| <-- CISCO >> <-- PFSENSE <-- SIPX SITEB >> >> Any thoughts on what the problem could be? >> >> I have bypassed everything on the firewall at the moment. >> >> Thank you in advance. >> >> Rhon >> >> >> > > > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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