You can try using "Transfer" in asterisk. Ive not used it in a while but I believe it construct a 302 (temporary moved) if used on incoming calls.
Stuff like that though as already pointed out your at the whim of your carriers security settings. On Fri, Jan 31, 2014 at 6:22 PM, <[email protected]> wrote: > 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] > > > > > > > > >
