No, a SIP reINVITE would still leave Branch A in the SIP path, but may take it out of the RTP path.
If you want it out of the SIP path as well, you'll want to look at SIP REFER, but I'm not sure if Unlimitel will allow the SIP call to be REFER'd to a URI with a different IP address. Further, I'm not even sure how you could accomplish that in the Asterisk dialplan, although I'm quite rusty on Asterisk these days - try the Transfer() dialplan command. -- Nabeel Jafferali X2 Networks Inc. On Fri, Jan 31, 2014 at 7:45 AM, Chad Osmond <[email protected]> wrote: > 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] > > > > >
