Hi Peter, Peter den Hartog wrote: > > Bogdan-Andrei Iancu wrote: > >> Hi Peter, >> >> Peter den Hartog wrote: >> >>> Hello, >>> >>> I don't know if i'm on the right mailing list for this issue but maby i'm >>> not the only one that had it :-). >>> >>> >> if it is opensips related, you are on the right list :) >> >>> I implemented opensips and it works good, the normal calls are going >>> great, >>> outside/inside it all works. inside transfer (exten to exten) works to. >>> >>> But when an outside caller calls the office, it goes to the asterisk, and >>> asterisk forwards it to an opensips extension. exten = >>> x,Dial,1,(SIP/[email protected]) That works great, the caller gets the >>> right >>> person, but when the one being called, transfer that call it gone. >>> >>> >> This is the first scenario where * is fronting OpenSIPS ...typically is >> the other way around :D >> >>> I think it's because asterisk is trying to transfer this caller, but the >>> extension is not there (it's in opensips ofcourse, but not in *) >>> >>> >> Normally, the call transfer (from the phone) is done via a REFER request >> (inside the ongoing dialog) - What I suspect is that , as * is in the >> path of all calls with external users, * will intercept the REFER and >> try to handle it locally. >> >> Try to get a trace and see if this is what happens = REFER being >> consumed by *, instead of passing it to the external party. >> >> Regards, >> Bogdan >> >>> I can connect the asterisk users to the opensips users by connecting the >>> database, but is this really needed? or is there another issue here? Do i >>> miss something? >>> >>> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > Hello Bogdan, > > That is correct, > in Asterisk i see nothing of a new call, or a transfer.. but the phone is > creating a new call on line 2, in opensips i just see a new ongoing call. > (the line 2 call) and on the outside phone i hear the asterisk wait/hold > music. > when doing call transfer via REFER, the REFER is propagating to the other party and the other party is responsible foe generating the new call - but as you have the Asteirsk on the path,it will behave as a end point, so * must generate the new call. > Is there any smart solution for this? can i just forward the complete call > to opensips and let asterisk only forward it, and not create the call? (it > now just does a dial to the sip member in opensips) > hmmm...not following here..could you detail a bit?
Regards, Bogdan _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
