I got rid of that error by putting allowguests=no in asterisk, then i don't get the error. i've put a number to a direct opensips extention with _xxxxxxxxx,Dial,1,(SIP/1...@172.16.1.14) (.14 beeing my opensips server :))
I called in again, and it ringed @ 105, that works great. Then i do a anounched transfer from 105 to 103, and again no succes. Again i added the log: http://dl.getdropbox.com/u/1382962/grep2.txt Peter den Hartog wrote: > > Yes, i just noticed that error myself, that's something else didn't had > that before today :-), but that's the whole issue, i think it's not > sending a new refer, it just creates a new call on line 2, and when i try > to press transfer and hang up the call disappears and in my phone screen i > see "transfer failed" > > the situation is like this: 104 is on Asterisk, 105 & 103 are on opensips, > 104 takes all the outside calls (for now i made it like this, so we are > able to transfer the calls announced) > > i call from my mobile, true the sip trunk to 104. I transfer a call from > 104 to 105, this works fine. Then i transfer the same call from 105 to > 103, these last 2 are both opensips extensions.. and that last part, > doesn't work. the ngrep of a call like this is what you can see in my last > post > > > Iñaki Baz Castillo wrote: >> >> El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió: >>> Ok thank you for your clear post, here is the grep of the call i made, >>> it's >>> a outside call to asterisk then a transfer to opensips. (anounched) >>> that >>> one is working, but then i try a transfer from 105 to 103, this is from >>> opensips extention to opensips extention. this one fails. >>> >>> http://dl.getdropbox.com/u/1382962/grep.txt >> >> I see no REFER request in that trace. >> Also what I see is a "484 Addres Incomplete" from Asterisk >> >>> Best regards >>> >>> Iñaki Baz Castillo wrote: >>> > El Lunes, 26 de Octubre de 2009, Peter den Hartog escribió: >>> >> Well yes, it does work for the internal calls, but >>> >> when a call comes in true asterisk to an opensips extention i CAN'T >>> >> transfer it :-), i get transfer failed in my screen of my phone, and >>> >> the call stays on the original called extention. This is only for >>> >> announced transfers, unannounced works fine. >>> >> >>> >> Flavio post stated something about routing your REFER's back to >>> >> asterisk, so it should work.. but i don't know how to route these >>> calls >>> >> back to the >>> >> asterisk. >>> > >>> > Please, you *already* have the answer. When a phone is speaking with >>> > Asterisk >>> > (through OpenSIPS) you must route REFER to Asterisk as *any* other >>> > in-dialog >>> > request, this is, the *same* as when a phone is speaking with other >>> phone >>> > directly (through OpenSIPS). >>> > >>> > If the REFER fails this is because Asterisk is rejecting it !!! >>> > >>> > I already suggested you to do a SIP capture (using ngrep) to inspect >>> > which error replies Asterisk when the REFER arrives to it. Please do >>> it >>> > and paste it >>> > here (I expect a 403 or 404, so it means a wrong configuration in you >>> > Asterisk, no more). >>> > >>> > And please, forget anything about exotic routing of the REFER. >>> >> >> >> -- >> Iñaki Baz Castillo <i...@aliax.net> >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > -- View this message in context: http://n2.nabble.com/Transfer-issue-tp3877950p3892280.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users