Peter den Hartog wrote: > > 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. >> >> 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) >> >> > > > Oke a little update, i can now do blind (cold) transfers from asterisk to > opensips (outside lines) but not hot transfers, then the call gets > disconnected. > Do you see some NOTIFY requests going around? they are used during attended transfer to inform on the new call state.....
Regards, Bogdan _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
