Gateways do not have to register fxo lines, only fxs ports register as users. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431
Email: [email protected] 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: Wen Jun <[email protected]> To: 'Tony Graziano' <[email protected]>; 'Wen Jun' <[email protected]> Cc: 'WORLEY, Dale R (Dale)' <[email protected]>; [email protected] <[email protected]> Sent: Tue Jun 08 09:29:41 2010 Subject: RE: [sipx-users] Call to FXO gateway of peer sipx server insite-to-site calling Got your point now.. Another stupid question which may be already asked about FXO gateway such as Audiocodes MP118, is it a must to let each FXO ports make registration to sipx ? Currently in my network I make each FXO ports registered in sipx and then make outbound and inbound PSTN call. -----Original Message----- From: Tony Graziano [mailto:[email protected]] Sent: Tuesday, June 08, 2010 5:08 PM To: Wen Jun Cc: WORLEY, Dale R (Dale); [email protected]; jun,wen Subject: Re: [sipx-users] Call to FXO gateway of peer sipx server insite-to-site calling I don't think you understand... If you are trying to use that gateway, you must set it up in your server as a gateway, going through the other server "will not work". Site A wants to use the gateway in site B. Site B's PSTN gateway has an IP address in 10.11.12.13. Site A sets up a gateway with ip address 10.11.12.13, and then configures a dialplan to send calls to it. You do not need to use "55" or some other code, that's up to you. Site A DOES NOT want to send the call to the sipx server in site B to for anything but site to site calling. To place a call to a gateway, the call should be sent directly to the gateway. The same gateway can exist on several sipx servers. On Mon, Jun 7, 2010 at 9:12 PM, Wen Jun <[email protected]> wrote: > Hi, Dale and Tony, > > Yes I've specified "55 + Any Number of digits" in my site-to-site > dialing plan. > > I made two call traces by "55 + remote internal extension" and "55 + 9 > + remote external PSTN". The call to remote internal extension was OK > but failed in the call to remote external PSTN. > > In failed call trace, I observed the call setup was stopped in the > "407 Proxy Authentication Required" by local sipx proxy to local > origination calling party. Local sipx proxy gives "Digest realm = > remote sipx domain rather than "local sipx domain" which leads local > origination calling party failed to Re-Invite with proper > authentication. In the well-done call trace to remote internal > extension, local sipx proxy gives "Digest realm = local sipx domain" > and then local origination calling party can make Re-Invite with proper authentication. > > Jun > > > -----Original Message----- > From: WORLEY, Dale R (Dale) [mailto:[email protected]] > Sent: Tuesday, June 08, 2010 7:05 AM > To: Wen Jun; [email protected] > Cc: 'jun,wen' > Subject: RE: [sipx-users] Call to FXO gateway of peer sipx server > insite-to-site calling > > ________________________________________ > From: [email protected] > [[email protected]] On Behalf Of Wen Jun > [[email protected]] > > I've well build site-to-site calling between to standalone sipx servers. > > I used suffix "55" to direct internal call from local endpoint to > remote endpoint of peer sipx site. In addition, I used suffix "9" to > lead call to local FXO gateway. > > I am trying to use suffix "559" to lead call to remote FXO gateway of > peer sipx servers, but that try is always failed. Is it supported in > sipx on this scenario ? > _______________________________________________ > > Without a trace of a failed call, it is impossible to know for sure > why the call failed. But I suspect that your dial plan rule for "55" > specifies "55 > + NN digits", where NN is the number of digits in your internal extensions. > If your dial plan rule says that, then "55 9 123-4567" will not match > the rule, as it doesn't have the correct number of digits. So you > would need to change the rule to "55 + any number of digits". > > In regard to "55", I would suggest that you assign each site a 2-digit > code (e.g., 55, 56, etc.), and have 55 route to its site from any > site's system (even from its own). That way, there is a way to dial > each extension that works from any site. > > Dale > > -- ====================== 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. _______________________________________________ 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/
