This is a sip reinvite isn't it? It should be a matter of the Branch A sending unlimitel and Branch B invites and taking itself out of the loop. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite
On Fri, Jan 31, 2014 at 12:06 AM, Dean Yorke <[email protected]> wrote: > Tough one. If they were with VoIP.ms you could use and attended > transfers and them just push it back with a code. > > > > Sent from my iPhone > > > On Jan 30, 2014, at 10:36 PM, Chuck Mariotti <[email protected]> > wrote: > > > > So, I have been asked to implement two asterisk boxes at two different > offices using different ISPs, etc... so I can't count on the connectivity > to be 100%. > > > > I have previously setup very basic SIP trunks between the offices (eg. > Ext 55xx is one office group and 44xx is the other office group... so you > can call extensions in either office, inbound DID calls in at one office, > can reach an extension in another office, etc...) In every case so far, the > need to do this has been minimal but has worked perfectly fine for the > situation. > > > > In this case however, it appears that DID number will be at one > office... and then passed to an extension (possibly at the other office). > The call volume is high enough that I am wondering if there is a way to > actually pass off the call to the remote asterisk box (and pass the SIP > connection to the DID provider along with it) so that the original > receiving server is eliminated from the packet route. Basically introduce > the two with a handshake and walk away. > > > > Can this be done easily? I imagine this is dependent on the DID > provider? Any hints as to what I should at? > > > > Their existing system is PBX in a Flash and Unlimitel trunks. > > > > Regards, > > > > Chuck > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [email protected] > For additional commands, e-mail: [email protected] > >
