Re: [OpenSER-Users] Broken "BYE" returned from Asterisk on TLS implementation ?

Hi, Arrggghh .. that's one of my attempts to eliminate the broken "BYE" problem... that's ngrep was captured when I set "modparam("rr", "enable_double_rr", "0");", I've paste another ngrep to http://pastebin.ca/674450, this time the double RR header is enabled. And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt the post is "openser"). Even though double RR header is enabled, but for BYE it's still doesn't process properly :( For the .cfg file line #130 onward, I did tried t_relay, forward and force_send_socket, but none of this will do the trick (force_send_socket was complaining TLS error due to missing certificate (?) ) Would appreciate if anyone could enlighten me why is this happen ? Thanks, David Loh Klaus Darilion wrote: > But the INVITE you posted at http://pastebin.ca/673392 also has > only one Record-Route header. > > regards > klaus > > David Loh schrieb: >> Hi, >> >> Yea, OpenSER proxy was add 2 record-route header for the >> INVITE/ACK ...but when asterisk disconnected the call and send BYE >> back to OpenSER, >> the TLS RR header wasn't present, the only 2 RR header was >> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" .... >> I'm puzzled ... is there any command to 'fix' this? >> >> >> Regards, >> David Loh >> >> Klaus Darilion wrote: >>> The openser proxy should add 2 record-route header (TLS and UDP = >>> double record route). This is why it does not work. >>> >>> regards >>> klaus >>> >>> David Loh schrieb: >>>> Hi All, >>>> >>>> Greeting. >>>> >>>> I've been struggle with OpenSER TLS implementation for more than >>>> a week, since I've ported from UDP to TLS, everything work fine >>>> except the "BYE" request from Asterisk (loose route), my >>>> implementation was something like below: >>>> >>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] >>>> >>>> My OpenSER.cfg already configured to listen on two port which is >>>> :- "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN >>>> (or even voicemail) having no problem, >>>> but when the callee disconnect the call, caller will never get >>>> hang up :( >>>> >>>> I've attached my ethereal trace/ngrep to pastebin, >>>> http://pastebin.ca/673392 >>>> >>>> Wondering if anyone can help me with the broken "BYE" that >>>> returned from Asterisk ? >>>> Line #131, supposedly this line should have contain 2 Via >>>> header, one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", >>>> but somehow the TLS via header was gone !! (compare to previous >>>> ACK (Line #117) /INVITE (Line #51). >>>> Due to the missing TLS via header, OpenSER log file was >>>> complaining "protocol/port mis-match". >>>> >>>> The last BYE request (Line #256) is actually firing from Client, >>>> which contain the "TLS" via. >>>> >>>> >>>> I've even tried "force_send_socket" to port 5061 (instead of >>>> 5060) from loose route, but it complaining TLS certificate error, >>>> since Asterisk doesn't support TLS natively, I've no clue why is >>>> the ACK/INVITE/CANCEL work but not BYE. >>>> if (loose_route) { >>>> .... >>>> if(is_method("BYE")) { force_send_socket(IP:5061); } >>>> } >>>> >>>> >>>> Has any one gone through of this kinda OpenSER over TLS + >>>> Asterisk setup, >>>> I'm really appreciate if you can share your experience with me, >>>> or pin point what's the mistakes I made here. >>>> >>>> Thanks in advance. >>>> >>>> Regards, >>>> David Loh >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > >