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]
