Do you have a SIP trunk between the two systems or a PSTN T1 trunk with 2 gateways?
If this is a SIP trunk, then users on system A cannot use a gateway on system B and get authorization to make a call. The reason is that users on system A do not have credentials for system B. Just calling a user on system B works because that does not require credentials. However, an outbound call does. The way around this would be to use a TLS connection between the two systems and set peer permissions accordingly. The TLS setup is described here: http://wiki.sipfoundry.org/display/xecsuserV4r2/Using+TLS If you use PSTN to connect the two then this should work as well. --martin > -----Original Message----- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of Danny Shay > Sent: Tuesday, September 21, 2010 5:04 PM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] Site-to-Site to PSTN > > I followed the wiki article to create the plan. > The only time it fails is if the user attempts to grab a pstn port on > the other end. > Tony, you helped me set it up properly when we initially rebuilt this > site. > > > Danny > > > -----Original Message----- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of Tony Graziano > Sent: Tuesday, September 21, 2010 3:35 PM > To: sipx-users@list.sipfoundry.org > Subject: Re: [sipx-users] Site-to-Site to PSTN > > Follow the wiki article on this. Using * "site-to-site" dial plan type > removes thje need for proxy authentication. > ============================ > 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: sipx-users-boun...@list.sipfoundry.org > <sipx-users-boun...@list.sipfoundry.org> > To: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> > Sent: Tue Sep 21 16:31:46 2010 > Subject: [sipx-users] Site-to-Site to PSTN > > I have a customer who has two Sipx boxes connected between a Point-to- > Point T1. > Users in at one branch can call the other branch's users all-day via > the site-to-site dial plan. > > At one time we were using Nortel SCS on both ends, and had the dial > plan set to send the entire matched suffix to the far-side pbx, so a > caller could dial local numbers at the other site. > 4 is the site-to-site dial plan prefix > 9 is the local calling dial plan prefix. > If the client would dial 495551212 the pbx would send "95551212" to the > far side pbx and it would go out as a local call. > > In the current build of Sipx this gives a "proxy authentication > required" > message (if I dial using x-lite softphone). Hard phones (LIP-68xx) > don't work either. > > Is this a bug or is it by design in the newer sipx versions? > > If this is the way it's supposed to be, what is the best practice > alternative? > I am currently working around the issue by having long distance dial > plans that match the far-side local prefixes automatically send through > the far side fxo gateway. Does this have any chance of confusing the > Sipx pbx that is actually "managing" the gateway, if the other gateway > is sending calls directly to it? > > Thank you in advance, > > Danny Shay > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
