I think it needs to be in the same sip domain (HA) in order for the AA on system A to locate and ring a phone at system B. The two systems should still have a unique extension range (for logic). I don't see an easy way to get the AA on one system to slave AA for a system on another, with or without permissions. If it is desirable to make it act like one system, it should be one system.
On Wed, Feb 16, 2011 at 9:30 PM, Josh M. Patten <[email protected]> wrote: > Tony I think he's referring to dialing extensions that are on system B from > an auto attendant on system A. System B is a completely different install > from system A. The problem I that the AA on system A has no way of knowing > what extensions are valid on system B. > > Not sure what could be done to solve this. > > Sent from my Samsung Moment™ only on the Now Network™ > ----- Original Message ----- > From:"Tony Graziano" <[email protected]> > To:"[email protected]" <[email protected]>, > "[email protected]" <[email protected]> > Cc:"[email protected]" <[email protected]> > Sent:2/16/2011 7:23 PM > Subject:Re: [sipx-dev] AA transfer to Site-To-Site Extensions > > > > You can do manual attended or blind transfers. > > You should be in the same sip domain so the aa at site A can transfer the > calls so it will know the handsets are "registered" to it. > > HOW you are trying to do this would require HA. > ============================ > 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: Sam Oredoyin <[email protected]> > To: sipXecs developer discussions <[email protected]> > Cc: Tony Graziano <[email protected]>; > [email protected] <[email protected]> > Sent: Wed Feb 16 20:06:55 2011 > Subject: Re: [sipx-dev] AA transfer to Site-To-Site Extensions > > Thanks Tony > > My setup is thus: > > SIPX 4.2.1: > 2 Servers, one has extension 1xxx and the other has extension 2xxx (Not HA) > Server A: (1xxx) is connected to the PSTN. > Server A has AA configured. > > When callers dial into the AA ( on server A) and then try to reach users in > the 2xxx range, the call does not go through and users get a response saying > the number is not valid. > > I have a site-to-site link between both servers. > > I hope this makes it clearer. > > Sam > > > > > On Thu, Feb 17, 2011 at 12:57 AM, Tony Graziano < > [email protected]> wrote: > >> I don't understand what you are saying. If you have 2 sipx systems with >> unique numbering plans, this will always work assuming your link between >> them and dns works properly. Please reference the wiki. >> ============================ >> 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: [email protected] >> <[email protected]> >> To: [email protected] <[email protected]>; >> sipXecs >> developer discussions <[email protected]> >> Sent: Wed Feb 16 19:54:13 2011 >> Subject: [sipx-dev] AA transfer to Site-To-Site Extensions >> >> I have looked for information on this for some time now. >> >> I have two phone systems that I would like to act as one. I would like >> the auto attendant on the sipxpbx to transfer calls to both systems, >> however it appears that the auto attendant can only transfer calls to >> the local INTERNAL extensions. I assume the auto attendant bypasses the >> dial plan. >> >> I need to remove the restriction from the AA, and yes I understand this >> would be a security risk. >> >> I will like to know where this can be done in the code (what files to edit >> etc) so that AA calls can go via the dial plan. >> >> Regards >> Sam >> _______________________________________________ >> sipx-dev mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ >> > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev/
