And the PBX of course needs to be able to resolve the far side SRV records / etc.
http://wiki.sipfoundry.org/display/xecsuserV4r2/sipXecs+to+sipXecs+Calli ng Mike > -----Original Message----- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of Tony Graziano > Sent: Tuesday, May 18, 2010 7:35 AM > To: c4rdi...@gmail.com > Cc: sipx-users@list.sipfoundry.org > Subject: Re: [sipx-users] No Voice/IVR on Site-to-Site > > I can't stress enough that for the gateway name, it is not a hostname, > it is > the domain (sip domain) name. > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: Tony Graziano <tgrazi...@myitdepartment.net> > To: Rhon <c4rdi...@gmail.com> > Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> > Sent: Tue May 18 01:23:25 2010 > Subject: Re: [sipx-users] No Voice/IVR on Site-to-Site > > 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 <c4rdi...@gmail.com> 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 > <mpic...@cmctechgroup.com > > > wrote: > > > >> It should be set to manual and yes. > >> > >> > >> > >> *From:* Rhon [mailto:c4rdi...@gmail.com] > >> *Sent:* Monday, May 17, 2010 9:33 AM > >> *To:* Picher, Michael; sipx-users@list.sipfoundry.org > >> *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 < > >> mpic...@cmctechgroup.com> wrote: > >> > >> Static NAT port on the pfSense? > >> > >> > >> > >> *From:* sipx-users-boun...@list.sipfoundry.org [mailto: > >> sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Rhon > >> *Sent:* Monday, May 17, 2010 9:14 AM > >> *To:* sipx-users@list.sipfoundry.org > >> *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 sipx-users@list.sipfoundry.org > > 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: tgrazi...@voice.myitdepartment.net > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpd...@voice.myitdepartment.net > 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. > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > 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/ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org 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/
